[svn-commits] lmadsen: tag 1.4.30-rc3 r250708 - /tags/1.4.30-rc3/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Mar 4 11:30:14 CST 2010
Author: lmadsen
Date: Thu Mar 4 11:30:10 2010
New Revision: 250708
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=250708
Log:
Importing files for 1.4.30-rc3 release.
Added:
tags/1.4.30-rc3/.lastclean (with props)
tags/1.4.30-rc3/.version (with props)
tags/1.4.30-rc3/ChangeLog (with props)
Added: tags/1.4.30-rc3/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.4.30-rc3/.lastclean?view=auto&rev=250708
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--- tags/1.4.30-rc3/ChangeLog (added)
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@@ -1,0 +1,27845 @@
+2010-03-04 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.30-rc3 released
+
+2010-03-03 21:28 +0000 [r250613] Leif Madsen <lmadsen at digium.com>
+
+ * doc/localchannel.txt: Update existing Local channel
+ documentation. A complete re-write of the Local channel
+ documentation has been performed, with the existing information
+ from localchannel.txt and localchannel.tex merged in. (issue
+ #16637) Reported by: kobaz Patches: localchannel.tex uploaded by
+ lmadsen (license 10) localchannel.txt uploaded by lmadsen
+ (license 10) Tested by: lmadsen, jsmith, mmichelson
+
+2010-03-03 19:04 +0000 [r250480] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Make sure to clear red alarm after
+ polarity reversal. From the issue: The automatic overnight line
+ tests (or manual ones) used on UK (BT) lines causes a red alarm
+ on a dahdi / TDM400P connected channel. This is because the line
+ uses voltage tests (battery loss) and polarity reversal. The
+ polarity reversal causes chan_dahdi to initiate v23 CallerID
+ processing but during this the event DAHDI_EVENT_NOALARM is
+ ignored so that the alarm is never cleared. (closes issue #14163)
+ Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded
+ by jedi98 (license 653) Tested by: mattbrown, Chainsaw,
+ mikeeccleston
+
+2010-03-03 18:02 +0000 [r250394] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: fixes problem with duplicate TXREQ packets
+ When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
+ call store_by_transfercallno() to link the chan_iax2_pvt struct
+ into iax_transfercallno_pvts. If a duplicate TXREQ packet is
+ received for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel
+
+2010-03-02 21:08 +0000 [r250041-250050] Leif Madsen <lmadsen at digium.com>
+
+ * doc/imapstorage.txt: Update IMAP documentation. Update the IMAP
+ documentation to make it clear that storing voicemails in the
+ same folder as a large number of emails could potentially cause
+ significant slow downs when writing or retrieving voicemails.
+ (closes issue #16704) Reported by: TimeHider Tested by: lmadsen,
+ TimeHider
+
+ * configs/cdr.conf.sample: Update documentation to clarify purpose
+ of unanswered option. (closes issue #16267) Reported by: elsto
+ Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
+ 10) Tested by: davidw, elsto
+
+ * doc/configuration.txt: Update documentation to not imply we
+ support overriding options. (issue #16855) Reported by: davidw
+
+2010-03-02 19:36 +0000 [r249845-249946] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * apps/app_echo.c: revert ability to exit echo app caused a
+ regression, as only supported VOICE, not VIDEO etc. Left in small
+ formatting change. (issue #16880)
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-03-01 19:35 +0000 [r249671] Sean Bright <sean at malleable.com>
+
+ * apps/app_voicemail.c: Fix crash in app_voicemail related to
+ message counting. We were passing a 'struct inprocess **' and
+ treating it like a 'struct inprocess *' causing a segfault.
+ (closes issue #16921) Reported by: whardier Patches:
+ 20100301_issue16921.patch uploaded by seanbright (license 71)
+ Tested by: whardier
+
+2010-03-01 17:02 +0000 [r249536] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c: Modify queued frames from local channels
+ to not set the other side to up In this case, attended transfers
+ were broken due to ast_feature_request_and_dial detecting the
+ channel being set to up before the answer frame could be read and
+ therefore failing to mark the channel as ready. This fix is a
+ regression fix for 244785, which should continue to work properly
+ as well. (closes issue #16816) Reported by: jamhed Tested by:
+ jamhed, corruptor
+
+2010-02-27 23:51 +0000 [r249365] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_dahdi.c: overlap receiving: automatically send CALL
+ PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
+ user has determined that sufficient call information has been
+ received the user shall stop T302 and send CALL PROCEEDING to the
+ network. Previously timeouts were possible if the dialplan took a
+ long time to issue any response back to the network. Verified
+ that our local TELCO also does the same. (issue #16789) Reported
+ by: alecdavis Patches: based on overlap_receiving_trunk.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis (closes
+ issue #16789)
+
+2010-02-27 14:07 +0000 [r249234] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_iax2.c: add a reference to the now-published IAX2
+ RFC
+
+2010-02-26 17:04 +0000 [r249100] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: For T.38 reINVITEs treat a 606 the same as a
+ 488. (closes issue #16792) Reported by: vrban Patches:
+ t38_606.patch uploaded by vrban (license 756)
+
+2010-02-25 21:22 +0000 [r248860] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location (again) This is an extension to 248757. As
+ such the dialplan test has been extended: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) exten => 5043, 1,
+ monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
+ changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
+ exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
+ changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
+ design and emits a warning exten => 5044, n, dial(sip/5001)
+
+2010-02-25 21:21 +0000 [r248859] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Some platforms clear /var/run at boot, which
+ makes connecting a remote console... difficult. Previously, we
+ only created the default /var/run/asterisk directory at install
+ time. While we could create it in the init script, that would not
+ work for those who start asterisk manually from the command line.
+ So the safest thing to do is to create it as part of the Asterisk
+ boot process. This also changes the ownership of the directory,
+ because the pid and ctl files are created after we setuid/setgid.
+ (closes issue #16802) Reported by: Brian Patches:
+ 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
+ Tested by: tzafrir
+
+2010-02-25 18:06 +0000 [r248668-248757] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location. Recordings should be placed in the monitor
+ directory when a non-absolute path is used. Exact dialplan used
+ for testing: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) ABE-2101
+
+ * apps/app_voicemail.c: Make deletion of temporary greetings work
+ properly with IMAP_STORAGE This same patch was merged in 220833,
+ but was skipped in this branch erroneously. (closes issue #16170)
+ Reported by: francesco_r
+
+2010-02-24 21:02 +0000 [r248582] Tilghman Lesher <tlesher at digium.com>
+
+ * main/logger.c: Remove color code sequences from verbose messages
+ that go to logfiles. (closes issue #16786) Reported by: dodo
+ Patches: logger2.patch uploaded by dodo (license 989) Tested by:
+ tilghman
+
+2010-02-23 16:26 +0000 [r248396] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes invite with replaces deadlock (closes
+ issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
+ uploaded by dvossel (license 671) Tested by: pwalker, dvossel
+
+2010-02-22 13:52 +0000 [r248268] Olle Johansson <oej at edvina.net>
+
+ * apps/app_meetme.c: Don't log to debug unless debug is turned on
+
+2010-02-20 22:25 +0000 [r248106] Olle Johansson <oej at edvina.net>
+
+ * main/rtp.c: Make sure we support RTCP compound messages with zero
+ reports
+
+2010-02-19 19:11 +0000 [r248012] Tilghman Lesher <tlesher at digium.com>
+
+ * main/loader.c, /: Backport crash fix from trunk to 1.4, whereby
+ 'core show gracefully' could crash Asterisk. (closes issue
+ #16470) Reported by: kjotte
+
+2010-02-19 17:18 +0000 [r247910] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Merged revision 247904 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
+ 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
+ consistent with other channel technologies. The processing of
+ DTMF tones on the receiving side of an ISDN channel is
+ inconsistent with the way it is handled in other channels,
+ especially DAHDI analog. This causes DTMF tones sent from an ISDN
+ phone to be doubled at the connected party. We are using the
+ following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
+ Option one is necessary because the asterisk DSP DTMF detection
+ is better than mISDN's internal DSP. Not as many false positives.
+ Option two is necessary to transmit DTMF tones end to end when
+ mISDN channels are connected to SIP channels with out of band
+ DTMF for example. The symptom is that DTMF tones sent by an ISDN
+ phone are doubled on the way through asterisk when two mISDN
+ channels are connected with a Local channel in between or if it
+ is bridged to an analog channel. The doubling of DTMF tones is
+ because DTMF is passed inband to asterisk by the mISDN channel
+ and passed out of band once again after the release of the DTMF
+ tone. Passing it inband is wrong. Neither an analog channel nor
+ SIP channel passes DTMF inband if configured to inband DTMF.
+ Analog and SIP channels filter out the DTMF tones because they
+ use the voice frames returned by ast_dsp_process. But chan_misdn
+ passes the unfiltered input voice frames instead. To overcome one
+ aspect of the problem, the doubling of DTMF tones when two mISDN
+ channels are directly bridged, someone made an 'optimization',
+ where in that case the DTMF tone passed out-of-band to the peer
+ channel is not translated to an inband tone at the transmit side.
+ This optimization is bad because it does not work in general. For
+ example, analog channels or mISDN channels when bridged through
+ an intermediary local channel will generate DTMF tones from
+ out-of-band information. Also, of course, it must not be done
+ when there is no inband DTMF available. This patch fixes the
+ issue. Now chan_misdn will filter the received inband DTMF signal
+ the same as other channel types. Another change included: No need
+ to build an extra translation path because ast_process_dsp does
+ it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
+
+2010-02-18 19:38 +0000 [r247651] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_features.c: Copy the calling party's account code to the
+ called party if they don't already have one. (closes issue
+ #16331) Reported by: bluefox Tested by: mnicholson
+
+2010-02-18 16:53 +0000 [r247502-247508] Leif Madsen <lmadsen at digium.com>
+
+ * README-SERIOUSLY.bestpractices.txt: Add additional link to best
+ practices document per jsmith.
+
+ * README-SERIOUSLY.bestpractices.txt (added): Add best practices
+ documentation. (issue #16808) Reported by: lmadsen (issue #16810)
+ Reported by: Nick_Lewis Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/507/
+
+2010-02-18 04:19 +0000 [r247422] Russell Bryant <russell at digium.com>
+
+ * Makefile, sounds/Makefile: Tweak argument handling for wget in
+ the sounds Makefile. 1) Fix the check to see if we are using wget
+ to not be full of fail. The configure script populates this
+ variable with the absolute path to wget if it is found, so it
+ didn't work. 2) Allow some extra arguments to be passed in for
+ wget. This is just a simple change to allow our Bamboo build
+ script to tell wget to be quiet and not fill up our logs with
+ download status output.
+
+2010-02-17 16:24 +0000 [r247168] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Make sure that when autofill is disabled that
+ callers not in the front of the queue cannot place calls. (closes
+ issue #16834) Reported by: kebl0155 Patches:
+ app_queue_no_autofill.v1.patch uploaded by kebl0155 (license 356)
+
+2010-02-15 23:42 +0000 [r246709] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Make the menuselect instructions correct by allowing
+ 'make menuselect' to actually solve dependency problems.
+ (Previously, it would fail out again with the same message about
+ running 'make menuselect', which was NOT at all helpful.)
+
+2010-02-12 23:30 +0000 [r246545] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: lock channel during datastore removal On channel
+ destruction the channel's datastores are removed and destroyed.
+ Since there are public API calls to find and remove datastores on
+ a channel, a lock should be held whenever datastores are removed
+ and destroyed. This resolves a crash caused by a race condition
+ in app_chanspy.c. (closes issue #16678) Reported by:
+ tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
+ tim ringenbach (license 540) Tested by: dvossel
+
+2010-02-12 18:52 +0000 [r246460] Jason Parker <jparker at digium.com>
+
+ * main/channel.c: Fix some silly formatting, and remove unnecessary
+ option_debug checks
+
+2010-02-10 17:44 +0000 [r246115] David Vossel <dvossel at digium.com>
+
+ * apps/app_queue.c: fixes random deadlock in app_queue with
+ use_weight during reload (closes issue #16677) Reported by:
+ tim_ringenbach Patches: app_queue_use_weight_deadlock.diff
+ uploaded by tim ringenbach (license 540)
+
+2010-02-10 13:37 +0000 [r245944] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample: Include examples of FILTER usage
+ in extension patterns where a "." may be a risk.
+
+2010-02-10 08:24 +0000 [r245909] Olle Johansson <oej at edvina.net>
+
+ * res/res_smdi.c: Make sure that res_smdi loads regardless of
+ configuration, since chan_dahdi depends on res_smdi
+
+2010-02-09 22:55 +0000 [r245792] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Fixes iaxs and iaxsl size off by one issue.
+ 2^15 = 32768 which is the maximum allowed iax2 callnumber.
+ Creating the iaxs and iaxsl array of size 32768 means the maximum
+ callnumber is actually out of bounds. This causes a nasty crash.
+ (closes issue #15997) Reported by: exarv Patches: iax_fix.diff
+ uploaded by dvossel (license 671)
+
+2010-02-08 20:39 +0000 [r245496] Jason Parker <jparker at digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2f.c: Remove reference of
+ documentation in source directory. People don't always build
+ Asterisk from source (distro packages, anybody?).
+
+2010-02-08 11:57 +0000 [r245422] Olle Johansson <oej at edvina.net>
+
+ * res/res_features.c: Res_features depends on res_adsi in 1.4
+
+2010-02-05 18:32 +0000 [r245044] Kevin P. Fleming <kpfleming at digium.com>
+
+ * contrib/firmware (removed), LICENSE: Remove contrib/firmware
+ directory as it is empty Remove explicit license for IAXy
+ firmware as it is no longer included in the tree
+
+2010-02-05 17:03 +0000 [r244926] Sean Bright <sean at malleable.com>
+
+ * main/asterisk.c: Update main copyright date.
+
+2010-02-04 23:20 +0000 [r244785] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c: Change channel state on local channels for
+ busy,answer,ring. Previously local channels channel state never
+ changed. This became problematic when the state of the other side
+ of the local channel was lost, for example during a masquerade.
+ Changing the state of the local channel allows for the scenario
+ to be detected when the channel state is set to ringing, but the
+ peer isn't ringing. The specific problem scenario is described in
+ 164201. Although this was noted on one of the issues, here is the
+ tested dialplan verified to work: exten =>
+ 9700,1,Dial(Local/*9700 at default&Local/#9700 at default) exten =>
+ *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+ exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+ *9700,n,Dial(SIP/5001) exten => #9700,1,Wait(1) ;1 works, 3 did
+ not exten =>
+ #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+ issue #14992) Reported by: davidw
+
+2010-02-01 23:13 +0000 [r244070-244242] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Backup and restore original textfile, for
+ prosthesis (gerund of prepend). Also, fix menuselect such that
+ changing voicemail build options correctly causes rebuild.
+ (closes issue #16415) Reported by: tomo1657 Patches:
+ prepention.patch uploaded by tomo1657 (license 484) (with
+ modifications by me to backport to 1.4)
+
+ * res/res_features.c: When a transferer hangs up during an attended
+ transfer BEFORE the transfer is answered, don't stop playing MOH.
+ (closes issue #16513) Reported by: litnimax Patches:
+ atxfer_moh_16513.patch uploaded by gknispel proformatique
+ (license 261) Tested by: litnimax
+
+ * main/channel.c, channels/chan_local.c: Revert previous chan_local
+ fix (r236981) and fix instead by destroying expired frames in the
+ queue. (closes issue #16525) Reported by: kobaz Patches:
+ 20100126__issue16525.diff.txt uploaded by tilghman (license 14)
+ 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: kobaz, atis (closes issue #16581)
+ Reported by: ZX81 (closes issue #16681) Reported by: alexr1
+
+2010-01-28 18:48 +0000 [r243862-243863] Leif Madsen <lmadsen at digium.com>
+
+ * BUGS: Oops, correct wrong link (https vs. http) in previous
+ commit.
+
+ * BUGS: Update location of bug tracker in documentation.
+
+2010-01-28 15:03 +0000 [r243779] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Fix a bogus third argument to
+ ast_copy_string().
+
+2010-01-27 20:35 +0000 [r243570-243691] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_queue.c: Revert 243570, I should have looked at this
+ closer. Will reopen the issue, but am leaving the review closed
+ as the change was pointless. (issue #16488)
+
+ * apps/app_queue.c: Extend announcement URL used with Queue from 80
+ chars to PATH_MAX. (closes issue #16488) Reported by: syspert
+ Patches: soundfilelen.pacth-2 uploaded by syspert (license 938)
+ Review: https://reviewboard.asterisk.org/r/475/
+
+2010-01-27 18:06 +0000 [r243486] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c: Use a safe list traversal while checking for
+ duplicate vars in pbx_builtin_setvar_helper.
+
+2010-01-26 23:55 +0000 [r243390] David Vossel <dvossel at digium.com>
+
+ * res/res_features.c: fixes bug with channel receiving wrong
+ privileges after call parking (closes issue #16429) Reported by:
+ Yasuhiro Konishi Patches: features.c.diff uploaded by Yasuhiro
+ Konishi (license 947) Tested by: dvossel
+
+2010-01-26 18:19 +0000 [r243258] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: Remove unnecessary code in ast_read as issue
+ 16058 has been fully solved now.
+
+2010-01-25 21:50 +0000 [r242852-242969] Tilghman Lesher <tlesher at digium.com>
+
+ * main/Makefile, pbx/Makefile: Err, and use the new menuselect
+ define, too.
+
+ * build_tools/cflags.xml, build_tools/menuselect-deps.in,
+ configure, configure.ac: Only rebuild parsers by an option in
+ menuselect
+
+ * configure, main/Makefile, configure.ac, pbx/Makefile: Restore
+ FreeBSD to able-to-compile-ish-mode
+
+2010-01-25 20:08 +0000 [r242850-242851] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c: Remove debugging that indeed should have been
+ gone before commit. Sorry.
+
+ * main/manager.c: Report error when writing to functions returns
+ error in AMI setvar action
+
+2010-01-25 05:42 +0000 [r242520-242728] Tilghman Lesher <tlesher at digium.com>
+
+ * main/Makefile, pbx/Makefile: Buildbot pointed out an error
+ (thanks, buildbot!)
+
+ * main/Makefile, pbx/Makefile: Oops, should have used CMD_PREFIX,
+ not ECHO_PREFIX, for the commands.
+
+ * main/Makefile: Make the build of the Asterisk expression parser
+ match that of the AEL parser.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/ael/ael_lex.c, pbx/Makefile, makeopts.in: Only rebuild bison
+ and flex source files on demand, if bison and flex are detected
+ by the configure script. Changed after discussion on the -dev
+ list about possible unnecessary build failures, due to
+ checkouts/untars causing these special source files to possibly
+ be newer than their resulting C files. This should additionally
+ ensure that nobody need learn about extra Makefile arguments to
+ ensure the proper files get rebuilt when changes are made to
+ these special source files.
+
+2010-01-22 21:44 +0000 [r242423] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/Makefile: Rebuild from flex, bison sources when necessary.
+ (issue #14629) Reported by: Marquis Patches:
+ 20100121__issue14629.diff.txt uploaded by tilghman (license 14)
+
+2010-01-22 09:19 +0000 [r242226] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Initialize notify_types to NULL
+
+2010-01-22 01:48 +0000 [r242142] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/cdr.c: Add Dialed Number Identifier (DNID) field to cdr.
+ Branch support, retains ABI, if backend CDR collector is adaptive
+ then database requires 'dnid' field to be added, otherwise no
+ functional changes. Reported by: alecdavis Tested by: alecdavis
+ Patch cdr_dnid.diff2.txt uploaded by alecdavis (license 585)
+ Review: https://reviewboard.asterisk.org/r/455/
+
+2010-01-21 15:25 +0000 [r241932] Sean Bright <sean at malleable.com>
+
+ * configure, configure.ac: Fix configure check for
+ PTHREAD_ONCE_INIT when manually adding -Wall to CFLAGS. (closes
+ issue #16666) Reported by: romain_proformatique
+
+2010-01-21 05:53 +0000 [r241765] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_math.c: Guard against division by zero.
+
+2010-01-20 20:00 +0000 [r241626] David Vossel <dvossel at digium.com>
+
+ * Makefile: fixes parsing error in Makefile. Some echo lines were
+ missing "; . Thanks to jparker for pointing out the problem.
+
+2010-01-20 14:12 +0000 [r241543-241544] Sean Bright <sean at malleable.com>
+
+ * pbx/pbx_spool.c: Modify fix for issue 16554 to be more inline
+ with what is already in trunk. I should have taken a closer look
+ at trunk/1.6.x, as this bug has already been fixed in a much more
+ simple manner, by just settings o->vars to NULL after the
+ ast_pbx_outgoing_* calls. (issue #16554) Reported by: mav3rick
+
+ * pbx/pbx_spool.c: Fix a memory leak in pbx_spool when using SetVar
+ in a call file. In pbx_spool, when we are freeing our 'outgoing'
+ struct, we weren't deallocating the ast_variable list we had
+ built from SetVars in a call file. Adding a call to
+ ast_variables_destroy in our deallocation routine works, but only
+ if the variables have not already been passed into
+ ast_pbx_outgoing_app() or _exten(), both of which take care of
+ destroying the variable list for us. (closes issue #16554)
+ Reported by: mav3rick Patches: issue16554_20100119.patch uploaded
+ by seanbright (license 71) Tested by: mav3rick
+
+2010-01-20 09:38 +0000 [r241458] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/pbx.c: Update CDR variables as pbx starts Allows CDR
+ variables added in cdr.c:set_one_cid to become visable during the
+ call, by executing ast_cdr_update() early in __ast_pbx_run. Based
+ on cdr_update.diff3.txt (issue #16638) Reported by: alecdavis
+ Patches: cdr_update.diff3.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis
+
+2010-01-19 17:41 +0000 [r241228] Jason Parker <jparker at digium.com>
+
+ * Makefile: Allow parallel make (-j) to work properly. 1.4 changes
+ are quite different from the others. (issue #16489) Reported by:
+ Chainsaw Tested by: qwell
+
+2010-01-19 17:22 +0000 [r241227] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_agent.c: Fix deadlock in agent_read by removing
+ call to agent_logoff. One must always lock the agents list lock
+ before the agent private. agent_read locks the private
+ immediately, so locking the agents list lock is not an option
+ (which is what agent_logoff requires). Because agent_read already
+ has access to the agent private all that is necessary is to do
+ the required hanging up that agent_logoff performed. (closes
+ issue #16321) Reported by: valon24 Patches: bug16321.patch
+ uploaded by jpeeler (license 325)
+
+2010-01-18 19:54 +0000 [r241015] Sean Bright <sean at malleable.com>
+
+ * main/config.c: Plug a memory leak when reading configs with their
+ comments. While reading through configuration files with the
+ intent of returning their full contents (comments specifically)
+ we allocated some memory and then forgot to free it. This doesn't
+ fix 16554 but clears up a leak I had in the lab. (issue #16554)
+ Reported by: mav3rick Patches: issue16554_20100118.patch uploaded
+ by seanbright (license 71) Tested by: seanbright
+
+2010-01-18 16:51 +0000 [r240891] David Vossel <dvossel at digium.com>
+
+ * Makefile: updated transmit_silence option documentation in
+ asterisk.conf This patch updates the transmit_silence option to
+ better document why the option exists, and what it affects.
+ Thanks to russell for providing the verbage for this update.
+
+2010-01-18 13:27 +0000 [r240768] Olle Johansson <oej at edvina.net>
+
+ * utils/Makefile: Fix muted compilation in 1.4 only
+
+2010-01-15 23:06 +0000 [r240547] Russell Bryant <russell at digium.com>
+
+ * Makefile: Fix a spelling error in the asterisk.conf sample.
+
+2010-01-15 20:52 +0000 [r240414] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Disallow leaving more than maxmsg
+ voicemails. This is a possibility because our previous method
+ assumed that no messages are left in parallel, which is not a
+ safe assumption. Due to the vmu structure duplication, it was
+ necessary to track in-process messages via a separate structure.
+ If at some point, we switch vmu to an ao2-reference-counted
+ structure, which would eliminate the prior noted duplication of
+ structures, then we could incorporate this new in-process
+ structure directly into vmu. (closes issue #16271) Reported by:
+ sohosys Patches: 20100108__issue16271.diff.txt uploaded by
+ tilghman (license 14) 20100108__issue16271__trunk.diff.txt
+ uploaded by tilghman (license 14)
+ 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: jsutton
+
+2010-01-14 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.29
+
+2010-01-08 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.29-rc1
+
+2010-01-07 20:14 +0000 [r238409-238411] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: fixes crash in "scheduled_destroy" in
+ chan_iax A signed short was used to represent a callnumber. This
+ is makes it possible to attempt to access the iaxs array with a
+ negative index. (closes issue #16565) Reported by: jensvb
+
+ * channels/chan_sip.c: Change in sip show channels display format
+ allowing more digits for CID (closes issue 0016459) Reported by:
+ Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins
+ (license 953)
+
+2010-01-06 21:41 +0000 [r238230] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_cdr.c: Revise documentation on disposition values to
+ the actual values used. (closes issue #16289) Reported by:
+ wdoekes
+
+2010-01-06 15:18 +0000 [r237697-238009] Russell Bryant <russell at digium.com>
+
+ * apps/app_mp3.c: Resolve a crash due to an ast_frame not being
+ fully initialized. (closes issue #16531) Reported by: john8675309
+ (closes SWP-615)
+
+ * main/utils.c: Change a NOTICE log message to DEBUG where it
+ belongs. (closes issue #16479) Reported by: alexrecarey (closes
+ SWP-577)
+
+2010-01-04 21:45 +0000 [r237318-237573] Tilghman Lesher <tlesher at digium.com>
+
+ * main/say.c: Bounds checking for input string (closes issue
+ #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt
+ uploaded by tilghman (license 14)
+
+ * main/pbx.c: Regression in issue #15421 - Pattern matching (closes
+ issue #16482) Reported by: wdoekes Patches:
+ astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
+ 20091223__issue16482.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes, tilghman
+
+ * main/pbx.c, res/res_agi.c, include/asterisk/channel.h: Add a flag
+ to disable the Background behavior, for AGI users. This is in a
+ section of code that relates to two other issues, namely issue
+ #14011 and issue #14940), one of which was the behavior of
+ Background when called with a context argument that matched the
+ current context. This fix broke FreePBX, however, in a post-Dial
+ situation. Needless to say, this is an extremely difficult
+ collision of several different issues. While the use of an
+ exception flag is ugly, fixing all of the issues linked is rather
+ difficult (although if someone would like to propose a better
+ solution, we're happy to entertain that suggestion). (closes
+ issue #16434) Reported by: rickead2000 Patches:
+ 20091217__issue16434.diff.txt uploaded by tilghman (license 14)
+ 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman
+ (license 14) Tested by: rickead2000
+
+ * channels/chan_local.c: It's also possible for the Local channel
+ to directly execute an Application. Reviewboard:
+ https://reviewboard.asterisk.org/r/452/
+
+2010-01-02 09:52 +0000 [r237135] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Release memory of the contact acl before
+ unloading module
+
+2009-12-30 21:57 +0000 [r236981] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_local.c: Don't queue frames to channels that have
+ no means to process them. (closes issue #15609) Reported by:
+ aragon Patches:
+ 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
+ tilghman (license 14) Tested by: aragon Review:
+ https://reviewboard.asterisk.org/r/452/
+
+2009-12-30 20:25 +0000 [r236890] Jeff Peeler <jpeeler at digium.com>
+
+ * utils/astman.c: Remove conflicting function definitions
+ (asterisk.h) so LOW_MEMORY compiles.
+
+2009-12-28 15:12 +0000 [r236509-236585] Sean Bright <sean at malleable.com>
+
+ * include/asterisk/threadstorage.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Try a test
+ compile to see if PTHREAD_ONCE_INIT requires extra braces. There
+ was conditional code (based on build platform) to optioinally
+ wrap PTHREAD_ONCE_INIT in braces that was removed since it is
+ fixed in newer versions of Solaris/OpenSolaris, but I am still
+ running into it on Solaris 10 x86 so add a configure-time check
+ for it.
+
+ * apps/app_meetme.c: Avoid a crash with large numbers of MeetMe
+ conferences. Similar to changes made to Queue(), when we have
+ large numbers of conferences in meetme.conf (1000s) and we use
+ alloca()/strdupa(), we can blow out the stack and crash, so
+ instead just use a single fixed buffer. (closes issue #16509)
+ Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
+ by seanbright (license 71) Tested by: seanbright
+
+2009-12-27 18:19 +0000 [r236433] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Turn on colors in the daemon,
+ since there's many requests for it on Ubuntu.
+
+2009-12-26 15:26 +0000 [r236357] Kevin P. Fleming <kpfleming at digium.com>
+
+ * sounds/Makefile: update to latest releases with zero uid/gid
+
+2009-12-23 15:21 +0000 [r236261] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Properly set T.38 attributes and don't
+ return before T.38 ports are configured when T.38 is found but no
+ audio stream is found. (closes issue #16318) Reported by:
+ bird_of_Luck Patches: t38-sdp-parsing-fix3.diff uploaded by
+ mnicholson (license 96), written by vrban and mnicholson Tested
+ by: vrban, mihaill
+
+2009-12-23 02:55 +0000 [r236184] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_agi.c: If EXEC only gets a single argument, don't crash
+ when the second is used. (closes issue #16504) Reported by:
+ bklang
+
+2009-12-22 16:58 +0000 [r236062] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes issue with p->method incorrectly set
+ to ACK It is possible for a second ACK to come in for a
+ retransmitted message. If an ack does not match an unacked
+ message in our queue, restore the previous p->method as this ACK
+ is completely ignored. (closes issue #16295) Reported by:
+ omolenkamp Patches: issue16295_v2.diff uploaded by dvossel
+ (license 671)
+
+2009-12-21 19:43 +0000 [r235940] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Change Monitor to not assume file to write to
+ does not contain pathing. 227944 changed the fname_base argument
+ to always append the configured monitor path. This change was
+ necessary to properly compare files for uniqueness. If a full
+ path is given though, nothing needs to be appended and that is
+ handled correctly now. (closes issue #16377) (closes issue
+ #16376) Reported by: bcnit Patches:
+ res_monitor.c-issue16376-1.patch uploaded by dant (license 670)
+
+2009-12-21 16:45 +0000 [r235821] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_features.c: Send parking lot announcement to the channel
+ which parked the call, not the park-ee. (closes issue #16234)
+ Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt
+ uploaded by tilghman (license 14)
+ 20091221__issue16234__1.4.diff.txt uploaded by tilghman (license
+ 14) Tested by: yeshuawatso
+
+2009-12-18 22:39 +0000 [r235652] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, configure.ac: Revise verbiage, per #asterisk-dev
+ discussion
+
+2009-12-18 22:29 +0000 [r235635] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, include/asterisk/cdr.h: Correct CDR dispositions
+ for BUSY/FAILED This patch is simple in that it reorders the
+ disposition defines so that the fix for issue 12946 works
+ properly (the default CDR disposition was changed to
+ AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set
+ in ast_call to ensure all CDR records are written. The side
+ effects of CDR changes are scary, so I'm documenting the test
+ cases performed to attempt to catch any regressions. The
+ following tests were all performed using 1.4 rev 195881 vs head
+ (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A
+ (Both SIP and features) A calls B A blind transfers to C Hangup C
+ (Both SIP and features) A calls B A attended transfers to C
+ Hangup C A calls B A attended transfers to C (SIP) C blind
+ transfers to A (features) Hangup A All of the test scenario CDRs
+ matched. The following tests were performed just with the patch
+ to ensure proper operation (with unanswered=yes): exten
+ =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w)
+ exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue
+ #16180) Reported by: aatef Patches: bug16180.patch uploaded by
+ jpeeler (license 325)
+
+2009-12-18 21:18 +0000 [r235572] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, configure.ac: Point to the typical missing package,
+ not the cryptic "termcap support".
+
+2009-12-17 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.28
+
+2009-12-09 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.28-rc1
+
+2009-12-09 19:58 +0000 [r233782-233879] Russell Bryant <russell at digium.com>
+
+ * main/loader.c: Fix breakage of the "module load <module>" CLI
+ command.
+
+ * main/loader.c, formats/format_ilbc.c, formats/format_vox.c,
+ include/asterisk/module.h, formats/format_pcm.c,
+ formats/format_h263.c, formats/format_g723.c,
+ formats/format_h264.c, formats/format_jpeg.c,
+ formats/format_g726.c, formats/format_gsm.c,
+ formats/format_g729.c, formats/format_sln.c,
+ formats/format_wav.c, formats/format_ogg_vorbis.c,
+ formats/format_wav_gsm.c: Set a module load priority for format
+ modules. A recent change to app_voicemail made it such that the
+ module now assumes that all format modules are available while
+ processing voicemail configuration. However, when autoloading
+ modules, it was possible that app_voicemail was loaded before the
+ format modules. Since format modules don't depend on anything,
+ set a module load priority on them to ensure that they get loaded
+ first when autoloading. This version of the patch is specific to
+ Asterisk 1.4 and 1.6.0. These versions did not already support
[... 27057 lines stripped ...]
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