[svn-commits] bebuild: tag certified-1.8.6-cert1 r364632 - /certified/tags/1.8.6-cert1/
SVN commits to the Digium repositories
svn-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.
Added:
certified/tags/1.8.6-cert1/.lastclean (with props)
certified/tags/1.8.6-cert1/.version (with props)
certified/tags/1.8.6-cert1/ChangeLog (with props)
Added: certified/tags/1.8.6-cert1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.6-cert1/.lastclean?view=auto&rev=364632
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--- certified/tags/1.8.6-cert1/ChangeLog (added)
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@@ -1,0 +1,33527 @@
+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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