[asterisk-commits] bebuild: tag certified-1.8.6-cert1 r364632 - /certified/tags/1.8.6-cert1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Apr 30 09:20:18 CDT 2012


Author: bebuild
Date: Mon Apr 30 09:20:13 2012
New Revision: 364632

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=364632
Log:
Importing files for 1.8.6-cert1 release.

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    certified/tags/1.8.6-cert1/.version   (with props)
    certified/tags/1.8.6-cert1/ChangeLog   (with props)

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Added: certified/tags/1.8.6-cert1/ChangeLog
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--- certified/tags/1.8.6-cert1/ChangeLog (added)
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+2012-04-30  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.6-cert1 released.
+
+2012-04-27 19:12 +0000 [r364261-364263]  Jason Parker <jparker at digium.com>
+
+	* configure: Re-run bootstrap, to fix configure script.
+
+	* apps/app_dahdiras.c, res/res_ael_share.c, apps/app_talkdetect.c,
+	  tests/test_pbx.c, formats/format_vox.c, tests/test_aoc.c,
+	  res/res_timing_kqueue.c, channels/chan_unistim.c,
+	  tests/test_heap.c, cdr/cdr_sqlite3_custom.c,
+	  Makefile.moddir_rules, apps/app_image.c, apps/app_chanisavail.c,
+	  tests/test_db.c, channels/chan_gtalk.c, res/res_config_sqlite.c,
+	  channels/chan_skinny.c, tests/test_locale.c,
+	  build_tools/embed_modules.xml, apps/app_minivm.c,
+	  channels/chan_alsa.c, res/res_config_ldap.c, cdr/cdr_odbc.c,
+	  channels/sip/include/sip.h, apps/app_jack.c,
+	  tests/test_amihooks.c, utils/utils.xml, apps/app_festival.c,
+	  tests/test_dlinklists.c, channels/chan_console.c,
+	  apps/app_getcpeid.c, tests/test_sched.c, channels/chan_oss.c,
+	  configs/features.conf.sample, tests/test_netsock2.c, Makefile,
+	  apps/app_macro.c, channels/chan_nbs.c, makeopts.in,
+	  sounds/Makefile, tests/test_poll.c, main/netsock2.c,
+	  cel/cel_pgsql.c, res/res_snmp.c, apps/app_dictate.c,
+	  tests/test_logger.c, apps/app_ices.c, cdr/cdr_radius.c,
+	  main/config.c, tests/test_func_file.c, build_tools/cflags.xml,
+	  tests/test_security_events.c, apps/app_setcallerid.c,
+	  funcs/func_pitchshift.c, tests/test_time.c, cdr/cdr_sqlite.c,
+	  funcs/func_frame_trace.c, tests/test_devicestate.c,
+	  tests/test_utils.c, apps/app_mp3.c, tests/test_astobj2.c,
+	  configs/sip.conf.sample, formats/format_jpeg.c,
+	  res/res_config_pgsql.c, res/res_adsi.c, CHANGES,
+	  apps/app_queue.c, tests/test_strings.c, utils/Makefile,
+	  channels/chan_usbradio.c, channels/chan_jingle.c,
+	  channels/chan_misdn.c, tests/test_skel.c,
+	  res/res_timing_pthread.c, channels/chan_h323.c,
+	  cel/cel_sqlite3_custom.c, apps/app_sms.c, apps/app_zapateller.c,
+	  res/res_fax_spandsp.c, main/asterisk.c,
+	  tests/test_substitution.c, build_tools/cflags-devmode.xml,
+	  apps/app_meetme.c, res/res_phoneprov.c, tests/test_event.c,
+	  apps/app_alarmreceiver.c, cdr/cdr_pgsql.c, cdr/cdr_csv.c,
+	  channels/chan_phone.c, res/res_smdi.c, tests/test_stringfields.c,
+	  funcs/func_shell.c, apps/app_amd.c, pbx/pbx_realtime.c,
+	  apps/app_url.c, apps/app_confbridge.c, apps/app_externalivr.c,
+	  apps/app_adsiprog.c, apps/app_nbscat.c, channels/chan_sip.c,
+	  tests/test_app.c, apps/app_waitforsilence.c, configure.ac,
+	  apps/app_morsecode.c, pbx/pbx_lua.c, UPGRADE.txt,
+	  tests/test_linkedlists.c, cdr/cdr_tds.c, apps/app_waitforring.c,
+	  tests/test_acl.c, pbx/pbx_dundi.c,
+	  contrib/scripts/get_ilbc_source.sh, cel/cel_radius.c,
+	  apps/app_dahdibarge.c, apps/app_readfile.c, /,
+	  tests/test_gosub.c, apps/app_test.c, res/res_jabber.c,
+	  agi/Makefile, apps/app_osplookup.c, main/features.c,
+	  res/res_timing_timerfd.c, pbx/pbx_ael.c, channels/chan_mgcp.c,
+	  res/res_ais.c, agi/agi.xml, tests/test_expr.c,
+	  tests/test_ast_format_str_reduce.c, cel/cel_tds.c: Merge changes
+	  for Certified Asterisk 1.8.6 branch. This branch existed
+	  elsewhere temporarily. These are the changes that were made
+	  there.
+
+	* / (added): Create branch for Certified Asterisk 1.8.6. Copied
+	  from the revision at which 1.8.6.0-rc1 was tagged. Changes will
+	  be merged shortly. This is being done for historical purposes.
+
+2011-10-17 19:00 +0000 [r341249]  Jason Parker <jparker at digium.com>
+
+	* /channels/chan_sip.c:
+	  Initialize variables before calling parse_uri If parse_uri was
+	  called with an empty URI, some pointers would be modified and an
+	  invalid read could result. This patch avoids calling parse_uri
+	  with an empty contact uri when parsing REGISTER requests.
+	  AST-2011-012 (closes issue ASTERISK-18668) ........ Merged
+	  revisions 341189 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-09-30 19:26 +0000 [r338230-338758]  Jason Parker <jparker at digium.com>
+
+	* /contrib/scripts/get_ilbc_source.sh:
+	  Merged revisions 336572 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011)
+	  | 7 lines Update get_ilbc_source.sh script to work again.
+	  Recently iLBC support in Asterisk has changed after the
+	  acquisition of GIPS by Google. More information about how this
+	  may affect you is available in a blog post at:
+	  http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
+	  ........
+
+	* /tests/test_security_events.c,
+	  /tests/test_stringfields.c,
+	  /tests/test_skel.c,
+	  /tests/test_time.c,
+	  /tests/test_locale.c,
+	  /tests/test_acl.c,
+	  /tests/test_devicestate.c,
+	  /tests/test_utils.c,
+	  /tests/test_aoc.c,
+	  /tests/test_astobj2.c,
+	  /tests/test_poll.c,
+	  /tests/test_amihooks.c,
+	  /tests/test_substitution.c,
+	  /tests/test_heap.c,
+	  /tests/test_expr.c,
+	  /tests/test_ast_format_str_reduce.c,
+	  /tests/test_gosub.c,
+	  /tests/test_logger.c,
+	  /tests/test_dlinklists.c,
+	  /tests/test_app.c,
+	  /tests/test_event.c,
+	  /tests/test_linkedlists.c,
+	  /tests/test_db.c,
+	  /tests/test_sched.c,
+	  /tests/test_netsock2.c,
+	  /tests/test_pbx.c,
+	  /tests/test_strings.c,
+	  /tests/test_func_file.c:
+	  Merged revisions 332176,338551 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r332176 | pabelanger | 2011-08-16 15:10:13 -0500 (Tue, 16 Aug
+	  2011) | 4 lines Flag test modules as 'core' Review:
+	  https://reviewboard.asterisk.org/r/1369/ ........ r338551 | qwell
+	  | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) | 1 line Test
+	  modules have a support level of core. ........
+
+	* /apps/app_dahdibarge.c,
+	  /apps/app_readfile.c,
+	  /apps/app_setcallerid.c,
+	  /cdr/cdr_sqlite.c: Disable
+	  deprecated modules from being built by default.
+
+	* /apps/app_macro.c: Merged
+	  revisions 338084 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r338084 | pabelanger | 2011-09-27 15:10:13 -0500 (Tue, 27 Sep
+	  2011) | 2 lines Upgrade app_macro to core ........
+
+	* /build_tools/cflags.xml,
+	  /channels/chan_usbradio.c,
+	  /build_tools/cflags-devmode.xml,
+	  /agi/agi.xml,
+	  /utils/utils.xml,
+	  /build_tools/embed_modules.xml,
+	  /tests/test_db.c,
+	  /tests/test_netsock2.c:
+	  Merged revisions 338227 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) |
+	  1 line Add support levels to non-module sections of menuselect
+	  (cflags, utils, etc). ........
+
+2011-09-26 16:13 +0000 [r337972]  Jason Parker <jparker at digium.com>
+
+	* /apps/app_meetme.c: Merged
+	  revisions 335714 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r335714 | pabelanger | 2011-09-13 16:30:18 -0500 (Tue, 13 Sep
+	  2011) | 4 lines Meetme should have 'core' support level (closes
+	  issue ASTERISK-18542) ........
+
+2011-09-23 20:18 +0000 [r337921]  Jason Parker <jparker at digium.com>
+
+	* /channels/chan_phone.c,
+	  /apps/app_osplookup.c,
+	  /funcs/func_frame_trace.c,
+	  /apps/app_minivm.c,
+	  /apps/app_mp3.c,
+	  /apps/app_confbridge.c,
+	  /res/res_config_ldap.c,
+	  /channels/chan_mgcp.c,
+	  /apps/app_jack.c,
+	  /apps/app_adsiprog.c,
+	  /apps/app_nbscat.c,
+	  /res/res_config_pgsql.c,
+	  /apps/app_festival.c,
+	  /apps/app_waitforsilence.c,
+	  /res/res_adsi.c,
+	  /pbx/pbx_lua.c,
+	  /channels/chan_console.c,
+	  /apps/app_getcpeid.c,
+	  /channels/chan_oss.c,
+	  /cdr/cdr_tds.c,
+	  /channels/chan_jingle.c,
+	  /formats/format_vox.c,
+	  /res/res_timing_pthread.c,
+	  /channels/chan_h323.c,
+	  /cel/cel_sqlite3_custom.c,
+	  /apps/app_sms.c,
+	  /pbx/pbx_dundi.c,
+	  /channels/chan_nbs.c,
+	  /cel/cel_pgsql.c,
+	  /cdr/cdr_sqlite3_custom.c,
+	  /apps/app_test.c,
+	  /apps/app_alarmreceiver.c,
+	  /apps/app_image.c,
+	  /apps/app_chanisavail.c,
+	  /apps/app_ices.c,
+	  /res/res_smdi.c,
+	  /funcs/func_pitchshift.c,
+	  /channels/chan_skinny.c,
+	  /pbx/pbx_ael.c,
+	  /pbx/pbx_realtime.c,
+	  /channels/chan_alsa.c,
+	  /apps/app_amd.c,
+	  /apps/app_url.c,
+	  /apps/app_externalivr.c,
+	  /cdr/cdr_odbc.c,
+	  /formats/format_jpeg.c,
+	  /res/res_ais.c,
+	  /cel/cel_tds.c,
+	  /apps/app_dahdiras.c,
+	  /apps/app_morsecode.c,
+	  /res/res_ael_share.c,
+	  /apps/app_talkdetect.c,
+	  /apps/app_waitforring.c,
+	  /channels/chan_misdn.c,
+	  /apps/app_zapateller.c,
+	  /res/res_fax_spandsp.c,
+	  /res/res_timing_kqueue.c,
+	  /channels/chan_unistim.c,
+	  /cel/cel_radius.c,
+	  /res/res_snmp.c,
+	  /apps/app_dictate.c,
+	  /res/res_phoneprov.c,
+	  /cdr/cdr_pgsql.c,
+	  /channels/chan_gtalk.c,
+	  /cdr/cdr_radius.c,
+	  /res/res_jabber.c,
+	  /res/res_config_sqlite.c,
+	  /cdr/cdr_csv.c: Disable
+	  extended/deprecated modules from being built by default.
+
+2011-09-13 15:54 +0000 [r335599-335601]  Jason Parker <jparker at digium.com>
+
+	* /main/features.c: Merged
+	  revisions 334840 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011)
+	  | 10 lines Fix AMI action Park crash. * Made AMI action Park not
+	  say anything to the parker channel (AMI header Channel2) since
+	  the AMI action is a third party parking the call. (This is a
+	  change in behavior that cannot be preserved without a lot of
+	  effort.) * Made not play pbx-parkingfailed if the Park 's' option
+	  is used. JIRA AST-660 ........
+
+	* /apps/app_queue.c: Merged
+	  revisions 333010 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011)
+	  | 12 lines Memory Leak in app_queue The patch that was committed
+	  in the 1.6.x versions of Asterisk for ASTERISK-15862 actually
+	  fixed two issues. One was not applicable to 1.8 but the other is.
+	  queue_leak.patch fixes the portion applicable to 1.8. (closes
+	  issue ASTERISK-18265) Reported by: Fred Schroeder Patches:
+	  queue_leak.patch (license #5049) patch uploaded by mmichelson
+	  Tested by: Thomas Arimont ........
+
+	* /UPGRADE.txt,
+	  /configs/sip.conf.sample,
+	  /CHANGES,
+	  /channels/sip/include/sip.h:
+	  Merged revisions 333009 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r333009 | mnicholson | 2011-08-23 13:11:50 -0500 (Tue, 23 Aug
+	  2011) | 11 lines default 'sipstorecause' to no We've decided to
+	  disable this feature by default in future 1.8 versions. This
+	  would be an unexpected behavior change for anyone depending on
+	  that SIP_CAUSE update in their dialplan. Please refer to the
+	  asterisk-dev mailing list more information:
+	  http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
+	  (issue AST-580) ........
+
+2011-08-23 21:58 +0000 [r332938-333068]  Jason Parker <jparker at digium.com>
+
+	* /apps/app_queue.c: Merged
+	  revisions 331774 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug
+	  2011) | 11 lines Unlock the channel before calling update_queue.
+	  Holding the channel lock when calling update_queue which attempts
+	  to lock the queue lock can cause a deadlock. This deadlock
+	  involves the following chain: 1. hold chan lock -> wait queue
+	  lock 2. hold queue lock -> wait agent list lock 3. hold agent
+	  list lock -> wait chan list lock 4. hold chan list lock -> wait
+	  chan lock ........
+
+	* /apps/app_queue.c: Merged
+	  revisions 332874 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011)
+	  | 18 lines Reference leaks in app_queue. * Fixed
+	  load_realtime_queue() leaking a queue reference when it
+	  overwrites q when processing a realtime queue. (issue
+	  ASTERISK-18265) * Make join_queue() unreference the queue
+	  returned by load_realtime_queue() when it is done with the
+	  pointer. The load_realtime_queue() returns a reference to the
+	  just loaded realtime queue. * Fixed queues container reference
+	  leak in queues_data_provider_get(). * queue_unref() should not
+	  return q that was just unreferenced. * Made logic in
+	  __queues_show() and queues_data_provider_get() when calling
+	  load_realtime_queue() easier to understand. ........
+
+	* /main/config.c: Merged
+	  revisions 332759 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011)
+	  | 15 lines Memory leak reading realtime database variable list.
+	  Calling ast_load_realtime() can leak the last list node if the
+	  read list only contains empty variable value items. * Fixed list
+	  filter loop in ast_load_realtime() to delete the list node
+	  immediately instead of the next time through the loop. The next
+	  time through the loop may not happen if the node to delete is the
+	  last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
+	  Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
+	  patch uploaded by rmudgett ........
+
+	* /res/res_timing_timerfd.c:
+	  Merged revisions 332320 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011)
+	  | 10 lines Don't read from a disarmed or invalid timerfd Numerous
+	  isues have been reported for deadlocks that are caused by a
+	  blocking read in res_timing_timerfd on a file descriptor that
+	  will never be written to. This patch adds some checks to make
+	  sure that the timerfd is both valid and armed before calling
+	  read(). Should fix: ASTERISK-1842, ASTERISK-18197,
+	  ASTERISK-18166, AST-486 AST-495, AST-507 and possibly others.
+	  ........
+
+	* /main/features.c,
+	  /CHANGES,
+	  /configs/features.conf.sample,
+	  /main/asterisk.c: Merged
+	  revisions 332100 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011)
+	  | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183
+	  Multi-parkinglot directs calls to wrong parkinglot. JIRA
+	  ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
+	  ParkedCall() with no extension should pickup first available call
+	  and does not. JIRA AST-576 Issues with parking lots * Removed
+	  searching for parking lots by extension. Parking lots can only be
+	  found by the parking lot name since parking lot access extensions
+	  and spaces are not guaranteed to be unique. * Added
+	  parking_lot_name option to the Park and ParkedCall applications.
+	  Updated documentation for Park and ParkedCall applications. * Add
+	  parkext_exclusive configuration option to make parking entry
+	  extensions specify which parking lot they access. (closes issue
+	  ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
+	  David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
+	  Quezada (closes issue ASTERISK-17430) Reported by: Philippe
+	  Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
+	  AST-624 'next' setting for findslot does nothing * Reimplemented
+	  since findslot feature option broken by -r114655. (closes issue
+	  ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
+	  JIRA ASTERISK-15792 Dialplan continues execution after transfer
+	  to park. This happens for DTMF attended transfer, DTMF blind
+	  transfer, and DTMF one-touch-parking if the party initiating
+	  these features also initiated the call. * Fixed the return code
+	  from the affected builtin features when parking a call. (closes
+	  issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
+	  rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
+	  the expected call when picking up a parked call. This is mostly a
+	  documentation problem. However, the option is not reset to the
+	  default when features.conf is reloaded. * Updated
+	  features.conf.sample documentation for courtesytone and
+	  parkedplay options. * Reset the parkedplay option to default when
+	  features.conf is reloaded. JIRA AST-615 AMI Park action followed
+	  by features reload results in orphaned channels in parking lot. *
+	  Reloading features.conf will not touch parking lots that have
+	  calls still parked in them. Reload again at a later time. Misc
+	  additional fixes: * Added unit test for parking lot dialplan
+	  usage checking. * Made update connected line when a parked call
+	  is retrieved from a parking lot. * Made retrieved parked call
+	  stop ringing or MOH depending upon how the call was waiting in
+	  the parking lot. * Made CLI "features show" indicate if the
+	  parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
+	  variable to allow dynamic parking lots to specify the parking lot
+	  access extension. * Made AMI ParkedCalls action ParkedCall events
+	  have a Parkinglot header. * Made AMI ParkedCalls action
+	  ParkedCallsComplete event have a Total header. * Fixed potential
+	  deadlock from AMI Park action holding channel locks while calling
+	  masq_park_call(). * Fixed several places where ast_strdupa() were
+	  used inside of loops. (Mostly fixed by refactoring the loop body
+	  into its own function.) * Fixed copy_parkinglot() copying too
+	  much from the source parking lot. Extracted the parking lot
+	  configuration settings into struct parkinglot_cfg. * Refactored
+	  courtesytone playing code to put the channel not playing the tone
+	  in autoservice. * Fix when pbx-parkingfailed is played that the
+	  other channel is put in autoservice if it exists. * Fixed
+	  parkinglot reference leak in parked_call_exec() error paths. *
+	  Fixed parkinglot_unref() use of parkinglot after it was unreffed.
+	  * Made destroy the struct ast_parkinglot parkings lock when done.
+	  * Refactored the features.conf parking lot configuration code to
+	  eliminate redundancy. * Fixed feature reload to better protect
+	  parking lots. * Fixed parking lot container reference leak in
+	  handle_parkedcalls(). * Fixed the total count in
+	  handle_parkedcalls(). Review:
+	  https://reviewboard.asterisk.org/r/1358/ ........
+
+	* /channels/chan_sip.c,
+	  /channels/sip/include/sip.h:
+	  Merged revisions 332026 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug
+	  2011) | 4 lines use DEFAULT_STORE_SIP_CAUSE to set the default
+	  value for the 'storesipcause' option AST-580 ........
+
+	* /channels/chan_sip.c,
+	  /configs/sip.conf.sample,
+	  /CHANGES: Merged revisions
+	  332021 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug
+	  2011) | 9 lines Added the 'storesipcause' option to sip.conf to
+	  allow the user to disable the setting of HASH(SIP_CAUSE,<chan
+	  name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan
+	  name>) on the channel carries a significant performance penalty
+	  because of the usage of the MASTER_CHANNEL() dialplan function.
+	  AST-580 ........
+
+	* /channels/chan_sip.c:
+	  Merged revisions 331867 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011)
+	  | 6 lines Fixes locking inversion issues present in the handling
+	  of the sip REFER method. (closes issue ASTERISK-18082) Reported
+	  by: James Van Vleet ........
+
+2011-08-31  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.6.0 Released.
+
+2011-08-25  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.6.0-rc3 Released.
+
+	------------------------------------------------------------------------
+	r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines
+
+	Fix installation into directories containing spaces.
+
+	This also vastly simplifies the logic in sounds/Makefile
+
+	(Closes issue ASTERISK-18290)
+	Reported by: Paul Belanger
+	Review: https://reviewboard.asterisk.org/r/1379/
+	------------------------------------------------------------------------
+
+2011-08-22  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.6.0-rc2 Released.
+
+	------------------------------------------------------------------------
+	r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines
+
+	Segfault in shell_helper in func_shell.c.
+
+	The return value of popen() was not checked for failure to open.
+
+	(closes issue ASTERISK-18109)
+	JIRA SWP-3633
+	Reported by: Michael Myles
+	Tested by: rmudgett
+	------------------------------------------------------------------------
+	r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 13 lines
+
+	Re-add support for spaces in pathnames, including now spaces in DESTDIR.
+
+	This was initially added to 1.8 prior to release, primarily to support the
+	standard paths on Mac OS X, but was partially reverted recently in Subversion,
+	due to the lack of support for spaces in DESTDIR.  This commit restores support
+	for the standard paths on Mac OS X, and also includes support for spaces in
+	DESTDIR.
+
+	(closes issue ASTERISK-18290)
+	Reported by: pabelanger
+
+	Review: https://reviewboard.asterisk.org/r/1326/
+	------------------------------------------------------------------------
+	r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 7 lines
+
+	Fix possible error on stringification of IPv4-mapped addrs
+
+	The FreeBSD netsock2 test has been failing for a while. We were
+	pasing sa->len to getnameinfo instead of sa_tmp->len.
+
+	ASTERISK-18289
+	------------------------------------------------------------------------
+
+2011-08-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.6.0-rc1 Released.
+
+2011-08-10 13:47 +0000 [r331315]  Kinsey Moore <kmoore at digium.com>
+
+	* main/manager.c: AMI action ModuleReload returns Error if Module:
+	  missing or empty An empty string was not being checked for
+	  properly causing identification of the module to be reloaded to
+	  fail and return an Error with message "No such module." (closes
+	  issue AST-616)
+
+2011-08-09 22:12 +0000 [r331248]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_iax2.c, apps/app_parkandannounce.c, main/pbx.c,
+	  channels/chan_sip.c, main/features.c: Misc minor items found in
+	  code. * Add some reentrancy protection in pbx.c when creating the
+	  contexts_table hash table. * Fix inverted test in chan_sip.c
+	  conditional code. * Fix uninitialized variable and use of the
+	  wrong variable in chan_iax2.c. * Fix test of return value in
+	  app_parkandannounce.c. Explicitly testing for -1 is bad if the
+	  function does not actually return that value when it fails. *
+	  Fixup some comments and add some curly braces in features.c.
+
+2011-08-09 16:13 +0000 [r331146]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooGkClient.c,
+	  addons/chan_ooh323.c: move ast_cond_signal for admitted call
+	  after all data filled/freed clear all log channels by pointed
+	  number not only first free allocated callToken in ooh323_answer
+
+2011-08-09 15:58 +0000 [r331142]  Jason Parker <jparker at digium.com>
+
+	* doc/asterisk.8: Regenerate asterisk man page from sgml.
+
+2011-08-08 20:52 +0000 [r331038]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_musiconhold.c: In-queue MOH stops after a periodic
+	  announcement If the seek value is past the end of file when
+	  resuming G.722 MOH, MOH will cease to function for the duration
+	  of the MOH session through all starts and stops until saved state
+	  is cleared. Adjusting the code to guarantee a single valid read
+	  (which is already assumed) fixes the bug. (closes issue
+	  ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/
+	  Tested-by: Jonathan Rose <jrose at digium.com>
+
+2011-08-04 20:29 +0000 [r330843]  Terry Wilson <twilson at digium.com>
+
+	* configure, configure.ac: Make libsrtp instructions more explicit
+	  when linking fails (closes issue ASTERISK-18139)
+
+2011-08-04 19:37 +0000 [r330827]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/ooCmdChannel.c,
+	  addons/ooh323c/src/ooGkClient.c: change gk client behaivour on
+	  rrq/grq failures to setup timers and next tries after timeout
+	  instead of complete failure in the ooh323 stack
+
+2011-08-03 15:14 +0000 [r330705-330762]  Kinsey Moore <kmoore at digium.com>
+
+	* main/Makefile: editing files in main/editline does not ensure
+	  rebuild of libedit.a When editing a source file in main/editline,
+	  the build system does not rebuild libedit.a and uses the already
+	  existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this
+	  problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes
+
+	* channels/chan_dahdi.c, channels/sig_analog.c: Call pickup broken
+	  for DAHDI channels when beginning with # The call pickup feature
+	  did not work on DAHDI devices for anything other than feature
+	  codes beginning with * since all feature codes in chan_dahdi were
+	  originally hard-coded to begin with *. This patch is also applied
+	  to chan_dahdi.c to fix this bug with radio modes. (closes issue
+	  AST-621) Review: https://reviewboard.asterisk.org/r/1336/
+
+2011-08-02 20:51 +0000 [r330648]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* res/res_jabber.c: Convert an error message to actually be
+	  helpful.
+
+2011-08-02 16:15 +0000 [r330575-330581]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Fixes crash in chan_iax2. Fixes crash in
+	  chan_iax2 resulting from an edge case in the way control frames
+	  are queued during calltoken negotiation is complete. (closes
+	  issue ASTERISK-17610) Reported by: mgrobecker
+
+	* channels/chan_sip.c: Optimization to buffer initialization fix.
+
+	* channels/chan_sip.c: Fixes uninitialized string buffer in log
+	  message. (closes issue ASTERISK-17200) Reported by: lmadsen
+
+2011-08-01 15:22 +0000 [r330433]  Kinsey Moore <kmoore at digium.com>
+
+	* main/say.c: Incorrect playback for Spanish in some circumstances
+	  When you say the time in spanish and it is 01:00 - 01:59 or 13:00
+	  - 13:59 you must use female pronunciation "1F". The function
+	  "say_date_with_format_es" does not take this in account. (closes
+	  ASTERISK-15016) Patch-by: Luis Jimenez
+
+2011-07-30 23:56 +0000 [r330368]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c: Remove some redundant locking code in
+	  ast_do_masquerade(). Also updated some comments.
+
+2011-07-30 15:25 +0000 [r330311]  Gregory Nietsky <gregory at distrotech.co.za>
+
+	* main/channel.c: prevent double masqurading channels when one is
+	  been hung up and deadlock avoidance is used. There is a race
+	  condition in ast_do_masquerade / ast_hangup (at least) Reported
+	  by me signed off by schmidts with input from David Vossel Review:
+	  https://reviewboard.asterisk.org/r/1323/
+
+2011-07-29 17:18 +0000 [r330203-330213]  Sean Bright <sean at malleable.com>
+
+	* formats/format_wav.c: Correct the check for O_RDONLY.
+
+	* formats/format_wav.c: Only write to wav files that were opened to
+	  be written to.
+
+2011-07-28 21:42 +0000 [r330107]  Terry Wilson <twilson at digium.com>
+
+	* main/term.c: Make console colors work for TERM=xterm-256color
+
+2011-07-28 17:04 +0000 [r330050]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c: Merged revisions 330033 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu,
+	  28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and
+	  outgoing call legs of a data call are using different formats:
+	  a-law, u-law. When the call is bridged, the media stream is run
+	  through translation to convert the media formats. The translation
+	  is bad for data calls. * Make incoming call that does not
+	  explicitly specify u-law or a-law use the DAHDI channel's default
+	  law. The outgoing call always uses the default law from the DAHDI
+	  channel. (closes issue ABE-2800) Patches:
+	  jira_abe_2800_companding.patch (license #5621) patch uploaded by
+	  rmudgett ..........
+
+2011-07-28 15:45 +0000 [r329994]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_sip.c: Fix a SIP transfer deadlock. The locking in
+	  this function is very scary. There are like 6 structs involved.
+	  (closes issue AST-470)
+
+2011-07-28 15:26 +0000 [r329991]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_fax.c: check for CONFIG_STATUS_FILE_INVALID when loading
+	  the res_fax config file Patch by: tzafrir Reported by: tzafrir
+	  (closes issue ASTERISK-18161)
+
+2011-07-28 11:34 +0000 [r329895]  Sean Bright <sean at malleable.com>
+
+	* channels/chan_sip.c: Make the output of Externhost in 'sip show
+	  settings' more consistent.
+
+2011-07-27 19:27 +0000 [r329782]  Leif Madsen <lmadsen at digium.com>
+
+	* apps/app_confbridge.c: Change support for ConfBridge() in 1.8 to
+	  Extended.
+
+2011-07-27 19:17 +0000 [r329767]  Sean Bright <sean at malleable.com>
+
+	* Makefile.moddir_rules: Explicitly sort the module list so that
+	  the menuselect lists are sorted. (closes issue ASTERISK-18141)
+	  Reported by: Richard Miller Patches: sort-order.diff uploaded by
+	  seanbright (License #5060) Tested by: leifmadsen
+
+2011-07-27 18:10 +0000 [r329709]  Jonathan Rose <jrose at digium.com>
+
+	* configs/indications.conf.sample: Fix New Zealand indications
+	  profile based on http://www.telepermit.co.nz/TNA102.pdf (closes
+	  issue ASTERISK-16263) Reported by: richardf Patches:
+	  nz-indications.patch uploaded by richardf (License #6015)
+
+2011-07-27 04:23 +0000 [r329613]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* cdr/cdr_odbc.c: Duration and billsec are swapped in high
+	  resolution time. Closes ASTERISK-18024 Patches:
+	  20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
+
+2011-07-26 14:04 +0000 [r329527-329529]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_voicemail.c: Changes sound file for prepend
+	  "then-press-pound" to "vm-then-pound" which is the same prompt,
+	  only it turned out "then-press-pound" was part of extra sounds.
+	  Also, vm is more appropriate anyway.
+
+	* main/app.c, apps/app_voicemail.c, include/asterisk/app.h,
+	  configs/voicemail.conf.sample: Fixes some voicemail forwarding
+	  behavior based around prepend mode. Formerly, prepend forwarding
+	  would have the user record a message with no useful prompt and an
+	  expectation for the user to push a button on the phone when
+	  finished recording. If a length of silence was detected instead,
+	  the recording would be canceled and the user would re-enter the
+	  voicemail forwarding menu. Subsequent time-outs in prepend
+	  recording would also bug out in the sense that they would write
+	  over the original message and get sent to the recipient
+	  regardless of whether they timed out or were accepted. This patch
+	  fixes this issue and adds a prompt which will be played after a
+	  timeout informing the user that they needed to press a button.
+	  Currently, the sound files that we have are somewhat inadquate
+	  for this, so after the call we simply have Allison say "Please
+	  try again. Then press pound." which actually relies on two
+	  separate sound files. Just one would be more appropriate.
+	  reporter: Vlad Povorozniuc Review:
+	  https://reviewboard.asterisk.org/r/1327/
+
+2011-07-25 19:49 +0000 [r329471]  Paul Belanger <pabelanger at digium.com>
+
+	* main/enum.c: Decrease verbose messages to debug, to help clean up
+	  CLI.
+
+2011-07-22 21:10 +0000 [r329144-329333]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/pbx.c: Fix memory leak in an allocation error path of
+	  handle_statechange(). * Make use buffer accessor function in
+	  handle_statechange() rather than directly accessing the struct
+	  member. * Make use less redundant loop construct for iterating
+	  over hints.
+
+	* main/pbx.c: Deadlocks dealing with dialplan hints during reload.
+	  There are two remaining different deadlocks reported dealing with
+	  dialplan hints. The deadlock in ASTERISK-17666 is caused by
+	  invalid locking order in ast_remove_hint(). The hints container
+	  must be locked before the hint object. The deadlock in
+	  ASTERISK-17760 is caused by a catch-22 situation in
+	  handle_statechange(). The deadlock is caused by not having the
+	  conlock before calling the watcher callbacks. Unfortunately,
+	  having that lock causes a different deadlock as reported in
+	  ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made
+	  handle_statechange() no longer call the watcher callbacks holding
+	  any locks that matter. * Made hint ao2 destructor do the watcher
+	  callbacks for extension deactivation to guarantee that they get
+	  called. * Fixed hint reference leak in ast_add_hint() if the
+	  callback container constructor failed. * Fixed hint reference
+	  leak in complete_core_show_hint() for every hint it found for CLI
+	  tab completion. * Adjusted locking in
+	  ast_merge_contexts_and_delete() for safety. * Added
+	  context_merge_lock to prevent ast_merge_contexts_and_delete() and
+	  handle_statechange() from interfering with each other. * Fixed
+	  ast_change_hint() not taking into account that the extension is
+	  used for the hash key. (closes issue ASTERISK-17666) Reported by:
+	  irroot Tested by: irroot JIRA SWP-3318 (closes issue
+	  ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA
+	  SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/
+
+	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Document
+	  parkinglot in chan_dahdi.conf.sample. * Document existing feature
+	  in chan_dahdi.conf.sample. * Remove some dead code related to the
+	  parkinglot option.
+
+	* apps/app_directed_pickup.c: Update PickupChan documentation. The
+	  PickupChan uses the ampersand as the argument separator. Was
+	  documented as: PickupChan(channel[,channel2[,...][,options]])
+	  Fixed documentation to:
+	  PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
+	  This is a continuation of ASTERISK-17494 for v1.8 and later.
+	  (closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
+	  pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
+	  by Erik Smith Tested by: Erik Smith
+
+	* main/features.c: Dialplan bridge() app mutex 'current_dest_chan'
+	  freed more times than we've locked! This appears to be a leftover
+	  from when ast_channel was converted to ao2 objects. Simply
+	  removed the extraneous unlock. (closes issue ASTERISK-17772)
+
+2011-07-20 21:20 +0000 [r329027]  Paul Belanger <pabelanger at digium.com>
+
+	* UPGRADE.txt: Asterisk now requires libpri 1.4.11+ for PRI
+	  support.
+
+2011-07-20 20:52 +0000 [r329012]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+	  Backport useful CLI "pri show channels" command to v1.8. The "pri
+	  show channels" command is useful for debuging to see if there are
+	  any stuck B channels. .......... r307964 | rmudgett | 2011-02-15
+	  15:42:55 -0600 (Tue, 15 Feb 2011) | 9 lines Add CLI "pri show
+	  channels" command. List the current mapping of DAHDI B channels
+	  to Asterisk channel names and which calls are on hold or
+	  call-waiting. Calls on hold or call-waiting are not associated
+	  with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 ..........
+	  r308205 | rmudgett | 2011-02-17 14:21:56 -0600 (Thu, 17 Feb 2011)
+	  | 1 line Add more verbage to CLI command 'pri show channels'
+	  usage. .......... r312579 | rmudgett | 2011-04-04 11:17:58 -0500
+	  (Mon, 04 Apr 2011) | 59 lines Change also updates 'pri show
+	  channels' command with the "chan idle" column to report if a
+	  channel is available for use.
+
+2011-07-20 20:16 +0000 [r328987]  Terry Wilson <twilson at digium.com>
+
+	* tests/test_netsock2.c: We can't guarantee an eth0 is present
+	  FreeBSD test fails on this case presumably because there is no
+	  eth0 on the test machine. Better to just remove this test for
+	  now.
+
+2011-07-20 19:00 +0000 [r328935]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Inband DTMF regression The functionality of
+	  inband DTMF in chan_sip relied upon
+	  ast_rtp_instance_dtmf_mode_get/set not working properly to avoid
+	  calling ast_rtp_instance_dtmf_begin/end on RTP streams with
+	  inband DTMF. According to documentation,
+	  ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
+	  never inband. This fixes the regression introduced in revision
+	  328823.
+
+2011-07-19 21:29 +0000 [r328878]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* sounds/Makefile, Makefile, Makefile.moddir_rules: Revert partial
+	  attempt at handling pathnames with spaces. Revision 299794
+	  attempted to improve the build system to be able to handle
+	  pathnames (primarily DESTDIR) with spaces in them, since this is
+	  common on some platforms (including Mac OSX). Unfortunately, the
+	  changes were incomplete and did not actually provide the desired
+	  behavior, and as a side effect the functionality that ensured

[... 32733 lines stripped ...]



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