[svn-commits] qwell: tag 1.4.2 r59054 - in /tags/1.4.2: .lastclean
.version ChangeLog
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Mon Mar 19 15:45:19 MST 2007
Author: qwell
Date: Mon Mar 19 17:45:18 2007
New Revision: 59054
URL: http://svn.digium.com/view/asterisk?view=rev&rev=59054
Log:
importing files for 1.4.2 release
Added:
tags/1.4.2/.lastclean (with props)
tags/1.4.2/.version (with props)
tags/1.4.2/ChangeLog (with props)
Added: tags/1.4.2/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.2/.lastclean?view=auto&rev=59054
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--- tags/1.4.2/ChangeLog (added)
+++ tags/1.4.2/ChangeLog Mon Mar 19 17:45:18 2007
@@ -1,0 +1,5660 @@
+2007-03-19 Jason Parker <jparker at digium.com>
+
+ * Asterisk 1.4.2 released.
+
+2007-03-19 22:29 +0000 [r59049] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * funcs/func_strings.c: Oops, this should have been a %d all along
+
+2007-03-19 15:52 +0000 [r59042] Joshua Colp <jcolp at digium.com>
+
+ * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
+ reported by ajohnson)
+
+2007-03-19 15:42 +0000 [r59040] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
+ via -dev list)
+
+2007-03-18 20:37 +0000 [r59037] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
+ code 0 (reported by qwerty1979)
+
+2007-03-18 16:36 +0000 [r59035] BJ Weschke <bweschke at btwtech.com>
+
+ * apps/app_followme.c: Don't return a non-zero return code if the
+ profile doesn't exist, to match what the documentation says it
+ already does. (#9307 Reported by kkiely)
+
+2007-03-16 16:12 +0000 [r58992] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_page.c: Wait for the async thread to exit when hanging
+ up all of the paged phones under all circumstances. (issue #9181
+ reported by PhilSmith)
+
+2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant <russell at digium.com>
+
+ * configs/sla.conf.sample: fix a couple SLA documentation
+ references
+
+ * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
+ (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
+ doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
+ doc/channelvariables.txt (added), doc/ael.txt (added),
+ doc/billing.tex (removed), build_tools/prep_tarball,
+ doc/callingpres.txt (added), doc/enum.txt (added),
+ doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
+ doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
+ doc/security.txt (added), doc/imapstorage.txt (added),
+ doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
+ doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
+ doc/iax.txt (added), doc/ael.tex (removed),
+ doc/channelvariables.tex (removed), doc/enum.tex (removed),
+ doc/security.tex (removed), doc/math.txt (added), Makefile,
+ doc/imapstorage.tex (removed), doc/privacy.tex (removed),
+ doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
+ (added), apps/app_voicemail.c, doc/cliprompt.txt (added),
+ doc/chaniax.txt (added), doc/app-sms.txt (added),
+ doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
+ doc/ices.txt (added), doc/dundi.tex (removed),
+ doc/linkedlists.txt (added), doc/queuelog.txt (added),
+ doc/extconfig.txt (added), doc/radius.txt (added),
+ doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
+ doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
+ (removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
+ doc/queuelog.tex (removed), doc/configuration.txt (added),
+ doc/asterisk-conf.txt (added), doc/sla.pdf (added),
+ doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
+ (added), doc/mp3.tex (removed), doc/configuration.tex (removed),
+ doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
+ doc/channels.txt (added), doc/ip-tos.tex (removed),
+ doc/extensions.txt (added), doc/queues-with-callback-members.txt
+ (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
+ doc/misdn.txt (added), doc/manager.txt (added),
+ doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
+ doc/billing.txt (added), doc/localchannel.txt (added),
+ doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
+ (added), doc/00README.1st (added): Making these documentation
+ changes in the 1.4 branch upset various people, so these chanes
+ will only be done in the trunk.
+
+ * build_tools/prep_tarball: Add the --pdf option to the usage of
+ rubber in prep_tarball
+
+ * Makefile, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
+ configure script checking for GTK2 and some additional Makefile
+ targets to support gmenuselect
+
+2007-03-15 23:52 +0000 [r58946] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
+ common syntax and update the resulting appdocs TeX file
+
+2007-03-15 23:24 +0000 [r58941] Russell Bryant <russell at digium.com>
+
+ * doc/asterisk.tex: add a link to the rubber homepage
+
+2007-03-15 23:11 +0000 [r58939] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_setcdruserfield.c, main/pbx.c,
+ apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
+ Expand deprecation warnings from simply warning on use to the
+ builtin documentation.
+
+2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant <russell at digium.com>
+
+ * doc/asterisk.tex, Makefile: Add Asterisk version information to
+ the generated PDF
+
+ * build_tools/prep_tarball: have prep_tarball attempt to build
+ asterisk.pdf
+
+2007-03-15 22:32 +0000 [r58933] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * funcs/func_realtime.c: Function works fine, but the documentation
+ is backwards.
+
+2007-03-15 22:25 +0000 [r58931] Russell Bryant <russell at digium.com>
+
+ * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
+ (added), doc/freetds.txt (removed), doc/odbcstorage.txt
+ (removed), configure, doc/sla.tex, doc/cygwin.txt (removed),
+ doc/model.txt (removed), doc/channelvariables.txt (removed),
+ doc/ael.txt (removed), doc/billing.tex (added),
+ doc/callingpres.txt (removed), doc/enum.txt (removed),
+ doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed),
+ doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
+ doc/security.txt (removed), doc/imapstorage.txt (removed),
+ doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added),
+ doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac,
+ doc/iax.txt (removed), doc/ael.tex (added),
+ doc/channelvariables.tex (added), doc/enum.tex (added),
+ doc/security.tex (added), doc/math.txt (removed), Makefile,
+ doc/imapstorage.tex (added), doc/privacy.tex (added),
+ doc/realtime.txt (removed), doc/dundi.txt (removed),
+ doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
+ (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
+ doc/ast_appdocs.tex (added), doc/realtime.tex (added),
+ doc/ices.txt (removed), doc/dundi.tex (added),
+ doc/linkedlists.txt (removed), doc/queuelog.txt (removed),
+ doc/extconfig.txt (removed), doc/radius.txt (removed),
+ doc/cliprompt.tex (added), doc/chaniax.tex (added),
+ doc/hardware.txt (removed), doc/mp3.txt (removed),
+ doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
+ (added), doc/queuelog.tex (added), doc/configuration.txt
+ (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf
+ (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added),
+ doc/h323.txt (removed), doc/mp3.tex (added),
+ doc/configuration.tex (added), doc/asterisk-conf.tex (added),
+ doc/jitterbuffer.txt (removed), doc/channels.txt (removed),
+ doc/ip-tos.tex (added), doc/extensions.txt (removed),
+ doc/queues-with-callback-members.txt (removed), doc/apps.txt
+ (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt
+ (removed), doc/manager.txt (removed), doc/jitterbuffer.tex
+ (added), doc/extensions.tex (added), doc/billing.txt (removed),
+ doc/localchannel.txt (removed),
+ doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt
+ (removed), doc/00README.1st (removed): Merge changes from
+ svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc
+ directory into a single LaTeX formatted document so that we can
+ generate a PDF, HTML, or other formats from this information. *
+ Add a CLI command to dump the application documentation into
+ LaTeX format which will only be include if the configure script
+ is run with --enable-dev-mode. * The PDF turned out to be close
+ to 1 MB, so it is not included. However, you can simply run "make
+ asterisk.pdf" to generate it yourself. We may include it in
+ release tarballs or have automatically generated ones on the web
+ site, but that has yet to be decided.
+
+2007-03-15 18:13 +0000 [r58923] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Don't assume that the pvt structure will
+ still exist after calling schedule_delivery as it may not. (issue
+ #9278 reported by fmachado)
+
+2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Some people like to put "limitonpeer"
+ instead of "limitonpeers" in their configuration. While we're at
+ it, support "limitonpeerz" and "limitonpeerssssss". (inspired by
+ issue #9172)
+
+ * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the
+ examples section
+
+ * doc/security.txt, /: Merged revisions 58896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) |
+ 3 lines Add a note to the security file that the Asterisk CLI and
+ log files may contain sensitive information, and that people
+ should keep this in mind. ........
+
+ * configs/sla.conf.sample, apps/app_meetme.c: By default, don't
+ attempt to do any CallerID handling at all with SLA because it is
+ known to not work properly in some situations. However, add an
+ option to enable it for those that would like to use it anyway.
+ The short story behind this is that to properly handle CallerID
+ with SLA, we need the ability to change the CallerID on an
+ existing call, and we are not ready to handle that.
+
+2007-03-14 01:47 +0000 [r58880] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * funcs/func_strings.c: Issue 9162 -
+ pbx_substitute_variables_helper assumes the buffer is initialized
+ to all zeroes. This fixes a case where it wasn't.
+
+2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Ensure that the blinky lights show that the
+ trunk stopped ringing when the trunk hangs up before a station
+ has answered it. (issue #9234, reported by francesco_r)
+
+ * configs/sla.conf.sample: fix the reference to the SLA
+ documentation
+
+2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
+ lines Issue #9229 - No port in request URI on register to non
+ default SIP ports (neelakantan) ........
+
+ * channels/chan_sip.c: Don't hangup the call on OK or errors on
+ MESSAGE and INFO inside of a dialog (like video update requests).
+
+ * channels/chan_sip.c: Issue #9251 - Clear From URI from user
+ attributes (tgrman)
+
+2007-03-12 16:52 +0000 [r58833] Joshua Colp <jcolp at digium.com>
+
+ * /: Blocked revisions 58832 via svnmerge ........ r58832 | file |
+ 2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines We can't
+ use the assembler version of fetchadd_int under Intel Macs.
+ (issue #9254 reported by darrell budic) ........
+
+2007-03-12 13:08 +0000 [r58825-58826] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 57034,57523,57753,58558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
+ 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
+ bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
+ 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
+ r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
+ 1 line fixed another place where the out_cause was hardcoded to
+ 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
+ Mar 2007) | 1 line we can free channel 31 as well, since we can
+ occupy it ........
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, channels/misdn/ie.c,
+ channels/misdn/isdn_msg_parser.c: added UU transceiving and
+ corect handling for rdnis
+
+2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Allow RFC2833 compensation to compensate for even
+ stupider implementations by queueing up the end frame at the
+ start, not the actual end. (issue #8963 reported by AndrewZ)
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Add
+ matchexterniplocally setting which only substitutes your
+ externip/externhost setting if it matches the localnet setting. I
+ know of at least two people who need opposite settings, so I made
+ it an option! (issue #8821 reported by kokoskarokoska)
+
+2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix a few more places in chan_iax2 where
+ the ast_frame used for receiving a frame was not properly
+ initialized. - Interpolating a frame when the jitterbuffer is in
+ use - decrypting a frame when IAX2 encryption is on - frames in
+ an IAX2 trunk
+
+ * apps/app_meetme.c: Make the compiler happy and initialize a
+ variable.
+
+ * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added):
+ Merge some updates to the SLA documentation. I plan to keep
+ working on this to explain all of the expected behavior with call
+ handling, configuration details for specific phones, and other
+ things. However, I got tired of doing it in plain text, so I
+ switched to using LaTeX. I have included the PDF version. I
+ haven't been able to get a nice looking plain text version out of
+ it yet, but I'm not terribly concerned since this is supposed to
+ be more of the manual, while the plain text sample configuration
+ file is the reference.
+
+2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_voicemail.c: Fix spelling of unavailable in voicemail
+ documentation. (issue #9248 reported by tensai)
+
+ * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
+ lines If we are unable to lookup the host in a c line we have to
+ abort, otherwise the previous data is gone and we will
+ (potentially) have no data when all is said and done. ........
+
+2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Hang up the channel that put the call on hold
+ in the event processing thread to avoid a race condition. Also,
+ if the station originated the call that it is putting on hold,
+ don't hang up the trunk if it was the only station on the call
+ and it is hanging up due to hold and not a normal hangup.
+
+ * channels/chan_zap.c: Add a missing break statement so that
+ handling the above event does not incorrectly destroy the
+ channel. (issue #9242, andrew)
+
+2007-03-08 21:33 +0000 [r58479] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_odbc.c: Fix segfault (Issue 9236)
+
+2007-03-08 20:54 +0000 [r58474] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Refactor hold handling a bit so that it does
+ not require keeping the call up when a call is put on hold.
+
+2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Make early SDP seeding even smarter! We have to check
+ codecs in the make_compatible function too. (issue #9221 reported
+ by marcelbarbulescu)
+
+ * main/dsp.c, /: Merged revisions 58388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
+ lines Only print out debug message if the definition that makes
+ the variables shows up was actually defined. (issue #9233
+ reported by serginuez) ........
+
+2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/http.c: this change was not needed; fclose() handles closing
+ the file descriptor already
+
+ * apps/app_meetme.c: fix a compiler warning, and overwriting 'res'
+ value
+
+ * main/http.c: fix two cases where HTTP session file descriptors
+ would not be closed
+
+2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant <russell at digium.com>
+
+ * channels/chan_zap.c, configure, configure.ac: If we receive
+ ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
+ tzafrir) Also, update the configure script to make sure that we
+ don't try to build chan_zap if the installed version of zaptel
+ does not include ZT_EVENT_REMOVED.
+
+ * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey
+ Moore) Merged revisions 58242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
+ 7 lines Fix a problem where the Asterisk channel name could be
+ that of the wrong IAX2 user for a call. This is because the first
+ step of choosing this name is to look for an IAX2 peer that
+ happens to have the same IP/port number that this call is coming
+ from and assuming that is it. However, this is not always
+ correct. So, I have made it change this name after authentication
+ happens since at that point, we have an exact match. ........
+
+2007-03-07 17:52 +0000 [r58240] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have)
+ at least one matching codec before attempting early bridge SDP
+ seeding. (issue #9221 reported by marcelbarbulescu)
+
+2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant <russell at digium.com>
+
+ * /: Blocked revisions 58167 via svnmerge ........ r58167 | russell
+ | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a
+ misplaced block of code in the 1.2 version of the patch to fix
+ issue #8977 ........
+
+ * main/manager.c, /: Merged revisions 58164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) |
+ 4 lines If the channels acquired using the manager Redirect
+ action are not up, then don't attempt to do anything with them.
+ It could lead to weird behavior, including crashes. (issue #8977)
+ ........
+
+2007-03-06 23:10 +0000 [r58121] Steve Murphy <murf at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
+ line Fix for 9220: Eyebeam cannot renew subscriptions for
+ presence info. Reason: re-SUBSCRIBE requests don't include Accept
+ headers, which the rfc says are optional (to put it tersely), (it
+ uses MAY), and luckily, the sip_pvt struct has the format info
+ stored, so we simply leave it if the format is set, and the
+ accept header null. ........
+
+2007-03-06 23:00 +0000 [r58119] Russell Bryant <russell at digium.com>
+
+ * configs/voicemail.conf.sample: Clarify the documentation of the
+ dialout and sendvoicemail options. (issue #9000, caio1982 and
+ serge-v)
+
+2007-03-06 20:37 +0000 [r58053] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
+ lines Change error message to proper message ........
+
+2007-03-06 18:01 +0000 [r58023] Russell Bryant <russell at digium.com>
+
+ * channels/chan_skinny.c: Return an error of transmit_response is
+ called without a session. (issue #9002)
+
+2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Since chan_iax2 does not support reception
+ of DTMF with duration ensure that it is set to 0 on the frame.
+ (issue #8521 reported by gdhgdh)
+
+ * apps/app_meetme.c: Don't create a listen channel and record the
+ conference unless the option is turned on. (issue #9204 reported
+ by francesco_r)
+
+ * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
+ lines Make create_dirpath use our standard for return values. -1
+ is failure, 0 is success. (issue #9205 reported by ballares)
+ ........
+
+2007-03-05 15:20 +0000 [r57826] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c, /: Merged revisions 57825 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
+ line Fixed a typo introduced via 9156 (either the gotos or their
+ doc strings are wrong) ........
+
+2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp <jcolp at digium.com>
+
+ * main/slinfactory.c: Don't allow a NULL pointer to reach
+ ast_frdup. (issue #9155 reported by cmaj)
+
+ * res/res_jabber.c: Don't reference a potentially NULL pointer.
+ (issue #9199 reported by klolik)
+
+ * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198
+ reported by edgreenberg)
+
+2007-03-03 15:31 +0000 [r57707] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2,
+ pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7:
+ Updated the regression tests
+
+2007-03-03 06:45 +0000 [r57649] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007)
+ | 2 lines Memory leak of a list, if call recording was abandoned
+ ........
+
+2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * main/say.c: submitted patch for Georgian language, issue 9010,
+ submitted by Alexander Shaduri
+
+2007-03-03 00:02 +0000 [r57591] Russell Bryant <russell at digium.com>
+
+ * configs/sla.conf.sample: add missing configuration template.
+ Thanks to Lacy Moore on asterisk-users for pointing this out\!
+
+2007-03-02 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.1 released.
+
+2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell at digium.com>
+
+ * configure, configure.ac: Update the check that is used to
+ determine whether zaptel transcoder support is present. The
+ interface has changed.
+
+2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
+ lines If a SIP message comes in and goes to a method handler that
+ requires additional values that may not be present then send back
+ an error. ........
+
+2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c, /: Merged revisions 57458 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
+ line further refinement in wording of goto documentation, as per
+ 9156, goto not proceeding to next instruction ........
+
+ * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes
+ right, but 9184 points out the problem-- the escape is removed by
+ pbx_config, and pbx_ael should also, before sending it down into
+ the pbx engine. Also, you have to insert it back in, if you are
+ generating extensions.conf code from the AEL.
+
+2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell at digium.com>
+
+ * main/file.c: Return the correct digit that interrupted the
+ stream. This fixes exiting the Background application when using
+ the m option. (issue #9176, mjagdis)
+
+ * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt,
+ include/asterisk/channel.h: Merge changes from
+ svn/asterisk/team/russell/sla_updates * Originally, I put in the
+ documentation that only Zap interfaces would be supported on the
+ trunk side. However, after a discussion with Qwell, we came up
+ with a way to make IP trunks work as well, using some things
+ already in Asterisk. So, here it is, this now officially supports
+ IP trunks. * Update the SLA documentation to reflect how to setup
+ IP trunks. * Add a section in sla.txt that describes how to set
+ up an SLA system with voicemail. * Simplify the way DTMF
+ passthrough is handled in MeetMe. * Fix a bug that exposed itself
+ when using a Local channel on the trunk side in SLA. The
+ station's channel needs to be passed to the dial API when dialing
+ the trunk. * Change a WARNING message to DEBUG in channel.h. This
+ message is of no use to users.
+
+2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 57317 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar
+ 2007) | 2 lines Don't even attempt to optimize things when a
+ proxy channel is involved. It will just explode in weird and
+ unexplaineable ways. (issue #9175 reported by
+ clegall_proformatique) ........
+
+2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support at transnexus.com>
+
+ * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.
+
+2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell at digium.com>
+
+ * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla
+ docs
+
+ * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes
+ from svn/asterisk/team/russell/sla_updates * Add support for
+ private hold. By setting "hold=private" for a trunk, only the
+ station that put the call on hold will be able to retrieve it
+ from hold. Also, by setting "hold=private" for a station, any
+ call that station puts on hold can only be retrieved by that
+ station.
+
+ * apps/app_meetme.c: Minor formatting change
+
+ * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from
+ svn/asterisk/team/russell/sla_updates * Add support for the
+ "barge=no" option for trunks. If this option is set, then
+ stations will not be able to join in on a call that is on
+ progress on this trunk.
+
+2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c, /: Merged revisions 57118 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
+ line a small documentation update, to reflect reality in the goto
+ doc strings, as per 9156, Goto does not proceed to next prio if
+ jump fails ........
+
+2007-02-28 18:57 +0000 [r57093] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb
+ 2007) | 2 lines Fix a few more issues with the agent logoff CLI
+ command. (issue #9123 reported by arbrandes) ........
+
+2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell at digium.com>
+
+ * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of
+ changes from svn/asterisk/team/russell/sla_updates * Add support
+ for station ring delays. Ring delays can be set globally for a
+ station or for specific trunks on the station. * Fix a few bugs
+ in existing code. * Restructure and Reorganize code to improve
+ readability and maintainability. * Improve formatting of the "sla
+ show (trunks|stations)" CLI commands.
+
+2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Picky compiler...
+
+ * apps/app_speech_utils.c: Better handle timeouts when the
+ individual speaks after everything has been played but before the
+ timeout ends.
+
+2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_ael.c: I was surprised that I had not yet downgraded
+ missing goto targets and macro call defs to a warning, in case
+ they are in extensions.conf; I rectified this problem. Also, A
+ goto in a macro to a target in a catch block was not being found;
+ I fixed this too; the cause was that I needed to treat catch
+ statements like an extension in the find_match code.
+
+2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell at digium.com>
+
+ * apps/app_voicemail.c: Fix voicemail email attachments. I missed
+ the conversion of one of the line endings and there was an extra
+ one where it should not have been. (issue #9128)
+
+2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky,
+ picky... show deprecation warning in application help, too
+ (reported via list)
+
+2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell at digium.com>
+
+ * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where
+ if a device was not specified in alsa.conf, then we just use the
+ system default, instead of creating our own default of hw:0,0.
+ (issue #9139)
+
+2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp at digium.com>
+
+ * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
+ lines Obey the clearglobalvars option in extensions reload (or
+ dialplan reload depending on your version). (issue #9146 reported
+ by ramonpeek) ........
+
+2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix a crash in my last change to
+ iax2_indicate(). (issue #9150)
+
+2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_record.c: Update app_record documentation to use new CLI
+ command, core show file formats. (issue #9151 reported by junky)
+
+ * main/pbx.c: Use ast_strlen_zero to see if the language and/or
+ context argument is not present for Background instead of just
+ checking if it is NULL. (issue #9141 reported by mjagdis)
+
+2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Do more complete locking of the
+ chan_iax2_pvt struct in the indicate callback. (Problem brought
+ up by Ben Smithurst on the asterisk-dev list)
+
+2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp at digium.com>
+
+ * main/asterisk.c: Allow both of the show version files and core
+ show file versions CLI commands to work. (issue #9135 reported by
+ mvanbaak)
+
+2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Move a comment to be in the correct struct.
+
+ * /: Blocked revisions 56729 via svnmerge ........ r56729 | russell
+ | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure
+ that lock.h is included in utils.c with AST_API_MODULE defined so
+ that the implementations will be properly included when the
+ AST_INLINE_API functions are not going to be inlined. (issue
+ #9124, festr) ........
+
+2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * main/channel.c, /: Merged revisions 56684 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007)
+ | 3 lines Issue 9130 - If prev is the last item on the channel
+ list, then evaluating additional conditions (e.g. name prefix)
+ will cause a NULL dereference. ........
+
+2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Make sure to set a speeddials parent on
+ creation. Don't crash if hold is pressed when no call is active.
+ Don't return in places that we shouldn't..
+
+2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: update to match zaptel 1.4 API change that
+ was committed a few minutes ago
+
+2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 56504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) |
+ 8 lines Fix up a couple more signal handlers to not do bad things
+ that could cause various undesirable results. The other day, I
+ made Asterisk deadlock by hitting Control-C because of a bad
+ signal handler. Now, signal handlers just set a flag and write to
+ an alert pipe for the flag to be handled. Then, there is another
+ thread that is monitoring for these flags. If being run in
+ console mode, it is just the main thread. If Asterisk is in the
+ background, a thread is created to do it. ........
+
+2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp at digium.com>
+
+ * main/sched.c: Change log notice to debug. It is possible for a
+ scheduled item to execute and be deleted at close to the same
+ time and unavoidable. If this happens this message creeps up.
+
+2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) |
+ 4 lines Don't destroy mutexes before unregistering all of the
+ entry points from the core. Also, fix a potential memory leak
+ from not destroying the locks for all of the possible call
+ numbers (about 32k of them). ........
+
+2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming at digium.com>
+
+ * build_tools/make_version_h: build special version strings for
+ AADK/S800i builds
+
+2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell at digium.com>
+
+ * apps/app_voicemail.c: The IMAP storage code uses the same code to
+ build the email that is used when voicemail is sent via email
+ using something like sendmail. In the patch from bug 8033 to fix
+ various IMAP storage problems, the line endings in the email file
+ were changed in the code from "\n" to "\r\n". However, this
+ breaks sending regular voicemail to email. So, this change
+ conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
+ enabled. (issue #9128, patch by jarjarbinks, modified by me to
+ not break IMAP storage)
+
+2007-02-22 23:25 +0000 [r56280] Joshua Colp <jcolp at digium.com>
+
+ * /: Blocked revisions 56279 via svnmerge ........ r56279 | file |
+ 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always
+ defer Agent logoff if any channels are up until they hang up.
+ (issue #9123 reported by arbrandes) ........
+
+2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell at digium.com>
+
+ * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c,
+ doc/sla.txt: Merge changes from team/russell/sla_updates. This
+ batch of changes to the SLA code does a few different things. * I
+ made the SLA code event driven instead of having to act in a lot
+ of busy loops while dialing things to wait for state changes.
+ This makes the code more efficient and readable at the same time.
+ * I have implemented a couple of new features. The first is
+ inbound trunk ringing timeouts. This is an option that defines
+ how long to let an incoming call on a trunk to ring. * I have
+ also implemented ring timeouts for stations. They may be
+ specified for the entire station, meaning it is how long to let
+ the station ring before giving up. You can also specify a ring
+ timeout for a specific trunk on a station. So, you can say that
+ you only want a specific station to ring 5 seconds if it is line1
+ ringing, but otherwise, there is no timeout.
+
+2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
+ lines Only change the original or clone channel if it's the
+ channel behind the proxy channel, not if it's just a regular
+ bridged channel. ........
+
+2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development <support at transnexus.com>
+
+ * doc/osp.txt: Update OSP documentation for v1.4.
+
+2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Move message from verbose to debug
+
+2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf at digium.com>
+
+ * sounds/Makefile: updated the sound tarball versions in Makefile
+
+2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Restructure a little bit of code to reduce
+ nesting. There is no functionality change here.
+
+ * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) |
+ 3 lines If we receive a frame that is not in any of the
+ negotiated formats, then drop it. (potentially issue #8781 and
+ SPD-12) ........
+
+2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp at digium.com>
+
+ * main/cli.c: Print out deprecation notice on usage output of CLI
+ commands. (issue #8925 reported by blitzrage)
+
[... 4864 lines stripped ...]
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