[asterisk-commits] lmadsen: tag 1.4.30-rc3 r250708 - /tags/1.4.30-rc3/

SVN commits to the Asterisk project asterisk-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:
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    tags/1.4.30-rc3/.version   (with props)
    tags/1.4.30-rc3/ChangeLog   (with props)

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+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

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