[svn-commits] bebuild: tag 1.8.6.0-rc1 r331313 - /tags/1.8.6.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Aug 10 08:03:58 CDT 2011
Author: bebuild
Date: Wed Aug 10 08:03:54 2011
New Revision: 331313
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=331313
Log:
Importing files for 1.8.6.0-rc1 release.
Added:
tags/1.8.6.0-rc1/.lastclean (with props)
tags/1.8.6.0-rc1/.version (with props)
tags/1.8.6.0-rc1/ChangeLog (with props)
Added: tags/1.8.6.0-rc1/.lastclean
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--- tags/1.8.6.0-rc1/ChangeLog (added)
+++ tags/1.8.6.0-rc1/ChangeLog Wed Aug 10 08:03:54 2011
@@ -1,0 +1,33029 @@
+2011-08-10 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.6.0-rc1 Released.
+
+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
+ that stale headers in the Asterisk 'include' directory were
+ removed got broken. In addition, the check for stale (and
+ possibly incompatible) modules in the Asterisk 'modules'
+ directory also got broken, and would never report any stale
+ modules. Users upgrading to this version or later versions would
+ then see unexpected module load errors. Since there are few users
+ who actually want to install Asterisk into paths that contain
+ spaces, and a proper fix for the build system would take many
+ hours, the best solution for now is to just revert the partial
+ solution.
+
+2011-07-19 17:57 +0000 [r328770-328823] Kinsey Moore <kmoore at digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, channels/chan_sip.c: RTP bridge away with
+ inband DTMF and feature detection When deciding whether Asterisk
+ was allowed to bridge the call away from the core, chan_sip did
+ not take into account the usage of features on dialed channels
+ that require monitoring of DTMF on channels utilizing inband
+ DTMF. This would cause Asterisk to allow the call to be locally
+ or remotely bridged, preventing access to the data required to
+ detect activations of such features. (closes 17237) Review:
+ https://reviewboard.asterisk.org/r/1302/
+
+ * apps/app_meetme.c: MeetMe requests a PIN twice in some
+ circumstances If a call to MeetMe includes both the dynamic(D)
+ and always request PIN(P) options, MeetMe will ask for the PIN
+ two times: once for creating the conference and once for entering
+ the conference. This behavior was introduced in rev 311616 when
+ adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
+ controlling PIN entry for joining a conference. (closes AST-601)
+ Review: https://reviewboard.asterisk.org/r/1305/
+
+2011-07-19 01:35 +0000 [r328716] Terry Wilson <twilson at digium.com>
+
+ * tests/test_linkedlists.c (added), include/asterisk/linkedlists.h:
+ Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the
+ element passed in if it isn't found. This commit also adds linked
+ list unit tests. Review: https://reviewboard.asterisk.org/r/1321/
+
+2011-07-18 20:47 +0000 [r328593-328663] Mark Murawki <markm at intellasoft.net>
+
+ * apps/app_dial.c: app_dial may double free a channel datastore
+ When starting a call with originate, and having the callee
+ channel run Bridge() on pickup, we will double free the
+ dialed_interface_info datastore, causing a crash. Make sure to
+ check if the datastore still exists before trying to free it.
+ (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested
+ by: Mark Murawski
+
+ * channels/chan_sip.c: If the sip private structure is null,
+ sip_setoption() will defref the null pointer and crash. Ideally,
+ sip_setoption shouldn't be called if there is a lack of a sip
+ private structure. But this will fix a crash. (closes issue
+ ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark
+ Murawski
+
+ * main/asterisk.c: Fixed invalid read and null pointer deref on
+ asterisk shutdown. In some cases when starting asterisk with -c
+ and hitting control-c to shutdown, there will be an invalid read
+ and null pointer deref causing a crash. (closes issue
+ ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark
+ Murawski, Kinsey Moore, Tilghman Lesher
+
+2011-07-18 07:10 +0000 [r328540] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * funcs/func_odbc.c: Typo
+
+2011-07-15 20:41 +0000 [r328446] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_macro.c, channels/chan_jingle.c, apps/app_dahdibarge.c,
+ apps/app_readfile.c, apps/app_setcallerid.c,
+ channels/chan_vpb.cc, apps/app_meetme.c, cdr/cdr_sqlite.c,
+ channels/chan_h323.c: Revert changes to defaultenabled state for
+ modules in Asterisk 1.8
+
+2011-07-15 19:22 +0000 [r328427] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c: small gk processing fixes: -
+ decrease for 1 second registration ttl for very low expirations
+ (some providers expire few earlier than TTL) - delete rrq and
+ registration expire timers on URQ received as we make new
+ registration.
+
+2011-07-14 23:12 +0000 [r328302] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Missing SIP pvt and channel unlock in
+ sip_set_rtp_peer(). Regression introduced by -r326144. Add
+ missing SIP pvt and channel unlock in sip_set_rtp_peer().
+
+2011-07-14 20:13 +0000 [r328209] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_image.c, res/res_http_post.c, formats/format_wav_gsm.c,
+ utils/stereorize.c, pbx/pbx_loopback.c, funcs/func_shell.c,
+ main/features.c, channels/chan_alsa.c, apps/app_externalivr.c,
+ formats/format_jpeg.c, res/res_speech.c, formats/format_gsm.c,
+ apps/app_milliwatt.c, formats/format_g719.c,
+ apps/app_saycounted.c, apps/app_fax.c, apps/app_echo.c,
+ funcs/func_math.c, channels/chan_agent.c, apps/app_dahdiras.c,
+ utils/astman.c, res/res_ael_share.c, apps/app_transfer.c,
+ apps/app_playback.c, res/res_config_curl.c, funcs/func_curl.c,
+ apps/app_waitforring.c, channels/chan_misdn.c, tests/test_skel.c,
+ addons/cdr_mysql.c, codecs/codec_ilbc.c, apps/app_zapateller.c,
+ apps/app_chanspy.c, apps/app_cdr.c, tests/test_substitution.c,
+ funcs/func_md5.c, utils/muted.c, tests/test_gosub.c,
+ funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c,
+ cdr/cdr_radius.c, formats/format_siren7.c,
+ apps/app_controlplayback.c, funcs/func_config.c, main/manager.c,
+ bridges/bridge_builtin_features.c, funcs/func_volume.c,
+ cdr/cdr_sqlite.c, funcs/func_aes.c, funcs/func_frame_trace.c,
+ tests/test_devicestate.c, res/res_agi.c, tests/test_astobj2.c,
+ apps/app_confbridge.c, apps/app_ivrdemo.c,
+ res/res_clioriginate.c, res/res_calendar_icalendar.c,
+ funcs/func_dialplan.c, funcs/func_db.c,
+ tests/test_ast_format_str_reduce.c, res/res_fax.c,
+ res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
+ apps/app_waituntil.c, channels/chan_console.c,
+ apps/app_getcpeid.c, apps/app_queue.c, funcs/func_global.c,
+ funcs/func_extstate.c, channels/chan_usbradio.c,
+ apps/app_flash.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
+ formats/format_g729.c, funcs/func_dialgroup.c, funcs/func_env.c,
+ res/res_timing_dahdi.c, funcs/func_strings.c,
+ res/res_calendar_caldav.c, apps/app_chanisavail.c,
+ formats/format_sln16.c, apps/app_ices.c, apps/app_exec.c,
+ bridges/bridge_multiplexed.c, cel/cel_odbc.c,
+ formats/format_pcm.c, pbx/pbx_ael.c, formats/format_h263.c,
+ cdr/cdr_manager.c, res/res_clialiases.c, funcs/func_sprintf.c,
+ tests/test_app.c, apps/app_softhangup.c, codecs/codec_g726.c,
+ apps/app_morsecode.c, utils/smsq.c, bridges/bridge_simple.c,
+ tests/test_sched.c, apps/app_talkdetect.c, apps/app_db.c,
+ res/res_calendar_ews.c, funcs/func_callcompletion.c,
+ tests/test_acl.c, funcs/func_cdr.c, utils/ael_main.c,
+ utils/streamplayer.c, res/res_calendar.c, cel/cel_radius.c,
+ channels/chan_vpb.cc, res/res_snmp.c, apps/app_dictate.c,
+ apps/app_authenticate.c, res/res_phoneprov.c, funcs/func_logic.c,
+ res/res_jabber.c, funcs/func_uri.c,
+ funcs/func_audiohookinherit.c, res/res_config_odbc.c,
+ funcs/func_odbc.c, res/res_realtime.c, codecs/codec_resample.c,
+ formats/format_h264.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ tests/test_amihooks.c, codecs/codec_lpc10.c, channels/chan_sip.c,
+ cdr/cdr_syslog.c, funcs/func_lock.c, res/res_adsi.c,
+ utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c,
+ formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c,
+ formats/format_g723.c, utils/extconf.c, tests/test_poll.c,
+ addons/chan_ooh323.c, cdr/cdr_sqlite3_custom.c,
+ funcs/func_module.c, apps/app_sayunixtime.c,
+ cdr/cdr_adaptive_odbc.c, res/res_smdi.c, tests/test_time.c,
+ apps/app_skel.c, funcs/func_srv.c, apps/app_amd.c,
+ pbx/pbx_realtime.c, apps/app_url.c, apps/app_dial.c,
+ apps/app_page.c, channels/chan_bridge.c, apps/app_privacy.c,
+ codecs/codec_speex.c, apps/app_disa.c, res/res_mutestream.c,
+ res/res_monitor.c, apps/app_macro.c, res/res_timing_kqueue.c,
+ res/res_fax_spandsp.c, channels/chan_unistim.c,
+ funcs/func_base64.c, addons/app_mysql.c,
+ channels/chan_multicast_rtp.c, apps/app_meetme.c,
+ utils/hashtest.c, res/res_musiconhold.c, apps/app_followme.c,
+ res/res_config_sqlite.c, cdr/cdr_csv.c,
+ tests/test_security_events.c, formats/format_ilbc.c,
+ funcs/func_enum.c, channels/chan_phone.c,
+ tests/test_stringfields.c, funcs/func_groupcount.c,
+ tests/test_locale.c, addons/chan_mobile.c, cdr/cdr_custom.c,
+ res/res_security_log.c, apps/app_parkandannounce.c,
+ apps/app_while.c, apps/app_jack.c, res/res_rtp_asterisk.c,
+ apps/app_nbscat.c, codecs/codec_a_mu.c, tests/test_dlinklists.c,
+ res/res_convert.c, pbx/pbx_lua.c, utils/astcanary.c,
+ channels/chan_oss.c, tests/test_strings.c, res/res_srtp.c,
+ cdr/cdr_tds.c, res/res_timing_pthread.c,
+ apps/app_directed_pickup.c, channels/chan_h323.c,
+ cel/cel_sqlite3_custom.c, apps/app_senddtmf.c,
+ funcs/func_callerid.c, addons/app_saycountpl.c, cel/cel_pgsql.c,
+ funcs/func_speex.c, apps/app_dahdibarge.c, channels/chan_local.c,
+ tests/test_logger.c, apps/app_record.c, apps/app_playtones.c,
+ bridges/bridge_softmix.c, apps/app_alarmreceiver.c,
+ channels/chan_iax2.c, res/res_pktccops.c,
+ res/res_rtp_multicast.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+ funcs/func_pitchshift.c, channels/chan_skinny.c,
+ apps/app_dumpchan.c, main/http.c, cdr/cdr_odbc.c,
+ utils/refcounter.c, res/res_calendar_exchange.c, res/res_ais.c,
+ codecs/codec_g722.c, tests/test_expr.c, funcs/func_timeout.c,
+ cel/cel_tds.c, formats/format_wav.c, formats/format_ogg_vorbis.c,
+ funcs/func_cut.c, apps/app_speech_utils.c, apps/app_sendtext.c,
+ funcs/func_channel.c, utils/hashtest2.c, pbx/pbx_config.c,
+ funcs/func_iconv.c, apps/app_mixmonitor.c, formats/format_g726.c,
+ res/res_odbc.c, apps/app_voicemail.c, tests/test_heap.c,
+ addons/format_mp3.c, formats/format_sln.c, apps/app_readexten.c,
+ apps/app_userevent.c, codecs/codec_gsm.c, channels/chan_gtalk.c,
+ cdr/cdr_pgsql.c, tests/test_func_file.c, apps/app_setcallerid.c,
+ apps/app_osplookup.c, cel/cel_manager.c, cel/cel_custom.c,
+ tests/test_utils.c, apps/app_minivm.c, apps/app_mp3.c,
+ res/res_timing_timerfd.c, apps/app_directory.c,
+ res/res_config_ldap.c, formats/format_siren14.c,
+ apps/app_adsiprog.c, res/res_config_pgsql.c, apps/app_read.c,
+ funcs/func_version.c, codecs/codec_alaw.c, agi/eagi-test.c,
+ res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c,
+ apps/app_forkcdr.c, funcs/func_blacklist.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, apps/app_stack.c, funcs/func_devstate.c,
+ apps/app_verbose.c, addons/res_config_mysql.c,
+ utils/check_expr.c, funcs/func_rand.c, apps/app_readfile.c,
+ codecs/codec_adpcm.c, apps/app_test.c, tests/test_event.c:
+ Introduce <support_level> tags in MODULEINFO. This change
+ introduces MODULEINFO into many modules in Asterisk in order to
+ show the community support level for those modules. This is used
+ by changes committed to menuselect by Russell Bryant recently
+ (r917 in menuselect). More information about the support level
+ types and what they mean is available on the wiki at
+ https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
+
+2011-07-14 19:21 +0000 [r328205] Jonathan Rose <jrose at digium.com>
+
+ * res/res_monitor.c: Monitor application arguments requirements
+ fixed. Monitor was requiring options in spite of no individual
+ option on Monitor being required. Review:
+ https://reviewboard.asterisk.org/r/1320/
+
+2011-07-13 18:46 +0000 [r328014] Richard Mudgett <rmudgett at digium.com>
+
+ * configs/features.conf.sample: Add ATXFER_NULL_TECH note in
+ features.conf.sample.
+
+2011-07-12 22:53 +0000 [r327950] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/manager.c: Correct double-free situation in manager output
+ processing. The process_output() function calls ast_str_append()
+ and xml_translate() on its 'out' parameter, which is a pointer to
+ an ast_str buffer. If either of these functions need to
+ reallocate the ast_str so it will have more space, they will free
+ the existing buffer and allocate a new one, returning the address
+ of the new one. However, because process_output only receives a
+ pointer to the ast_str, not a pointer to its caller's variable
+ holding the pointer, if the original ast_str is freed, the caller
+ will not know, and will continue to use it (and later attempt to
+ free it). (reported by jkroon on #asterisk-dev)
+
+2011-07-12 20:07 +0000 [r327890] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_directory.c: search in the current context for 'a' and
+ 'o' instead of 'default'
+
+2011-07-12 19:38 +0000 [r327888] Jason Parker <jparker at digium.com>
+
+ * Makefile: Fix uninstall target, so that modules dir gets cleared
+ again.
+
+2011-07-12 19:10 +0000 [r327852] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c: Added additional checks for mailbox /
+ password beginning with '*' character A bug existed such that if
+ a user entered a password with '*', and the extension 'a' did not
+ exist, an invalid mailbox would be created and the user
+ authenticated. The code was changed to prevent this from
+ occurring, and to prevent users from having mailboxes or
+ passwords defined that begin with the '*' character. (closes
+ issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/
+
+2011-07-12 15:35 +0000 [r327793] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * tests/test_substitution.c: Use 'printf' (POSIX issue 4) instead
+ of 'echo -n', for portability. The problem with using 'echo -n'
+ is that it is not portable. While BSD systems required that the
+ '-n' option be removed and interpreted, System V required that
+ all strings should be echoed with no interpretation of options.
+ This fundamental difference of behavior means that it is never
+ possible to use the '-n' flag to echo in tests which are meant to
+ be portable. In this case, on Mac OS X 10.6, the /bin/sh shell
+ builtin 'echo' uses the System V semantics of the command, and
+ thus the SHELL test failed on that platform.
+ http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
+
+2011-07-11 19:41 +0000 [r327682] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/jingle.h, channels/chan_gtalk.c: Update
+ chan_gtalk to work with changed GMail-based calls The messages
+ sent by the GMail client have changed, but include the old-style
+ messages as well. This patch checks for this case and uses the
+ old-style offer. (closes issue ASTERISK-18084) Review:
+ https://reviewboard.asterisk.org/r/1312/
+
+2011-07-11 13:53 +0000 [r327512] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, tests/test_substitution.c: reset our buffer each
+ iteration when doing variable substitution
+
+2011-07-11 10:56 +0000 [r327411-327412] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * main/Makefile: Properly building the Debian armhf (HardFloat)
+ port. Remove the line that should have been removed in r327411.
+
+ * main/Makefile: fix building the Debian armhf (HardFloat) port
+ Fixes
+ http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385
+ (Missing pthreads)
+
+2011-07-08 22:27 +0000 [r327258] Jason Parker <jparker at digium.com>
+
+ * main/db1-ast/mpool, addons, cdr, formats, codecs/gsm/src, funcs,
+ addons/ooh323c/src, bridges, codecs/lpc10, main/db1-ast/btree,
+ codecs/g722, main, main/db1-ast/recno, channels/sip, res, pbx,
+ res/ael, channels, main/stdtime, addons/ooh323c/src/h323, codecs,
+ utils, main/db1-ast/hash, cel, apps, main/db1-ast/db: Add .o
+ files to svn:ignore property, since it's only ignored if locally
+ configured to do so.
+
+2011-07-08 21:41 +0000 [r327211] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: INVITE 403 Forbidden response always
+ retransmits the maximum times. Asterisk sends a 403 Forbidden
+ response if authentication fails for an INVITE as required.
+ However, it ignores the ACK and keeps retransmitting the
+ response. * Made not delete the to-tag in the dialog so the
+ expected ACK can be matched with the dialog and stop the
+ retransmissions.
+
+2011-07-08 19:52 +0000 [r327106] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, tests/test_substitution.c: Reset our ast_str before
+ passing it on to dialplan function backends. It is possible for a
+ dialplan backend to not modify the given buffer or ast_str and
+ still return success. This causes any previous value stored in
+ the buffer to be used as if the new function call provided it.
+ Some functions also append to the given buffer assuming it is
+ empty. The test_substitution unit test has also been modified to
+ detect this problem. (closes issue ASTERISK-17878)
+
+2011-07-08 16:00 +0000 [r327044-327046] Russell Bryant <russell at digium.com>
+
+ * tests/test_netsock2.c: Fix an error and add more log message info
+ to help see why this fails on FreeBSD.
+
+ * channels/chan_dahdi.c: Resolve some set-but-unused-variable
+ warnings.
+
+2011-07-08 01:08 +0000 [r326985] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c: Some code cleanup in pbx.c * Mostly comment and
+ format changes. * ast_context_remove_extension_callerid() and
+ ast_add_extension_nolock() will write lock the found specific
+ context. * ast_context_find() will now tolerate a NULL name. *
+ Eliminated some inlined versions of find_context() and
+ find_context_locked().
+
+2011-07-07 19:17 +0000 [r326830] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * res/res_http_post.c: libgen.h is also needed on Darwin for
+ basename(3)
+
+2011-07-07 16:04 +0000 [r326689] Jonathan Rose <jrose at digium.com>
+
+ * res/res_config_odbc.c: res_odbc patch by tilghman to fix integers
+ with null values Addresses some improper sql statements in
+ res_odbc that would cause an update to fail on realtime peers due
+ to trying to set as "(NULL)" rather than an actual NULL. (closes
+ issue #1922STERISK-17791) Reported by: marcelloceschia Patches:
+ 20110505__issue19223.diff.txt uploaded by tilghman (license 14)
+
+2011-07-07 15:28 +0000 [r326681-326683] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: use sips: or sip: depending on the transport
+ in use when building reply digest URIs
+
+ * channels/chan_sip.c: make the uri parameter used in reply digests
+ more standards compliant in certain cases by prepending "sip:" or
+ "sips:" to it
+
+2011-07-06 15:26 +0000 [r326484] David Vossel <dvossel at digium.com>
+
+ * res/res_timing_timerfd.c: Reverts fix for timerfd locking issue.
+ jrose discovered a performance issue with this fix that prevents
+ his analog phones from working when using timerfd as a timing
+ source. Until it is understood what is causing this performance
+ problem, this patch is being reverted.
+
+2011-07-06 14:35 +0000 [r326411-326469] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
+ channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_jingle.c, channels/chan_dahdi.c,
+ funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
+ funcs/func_aes.c: Removing type attributes, as a change to
+ menuselect makes them no longer necessary.
+
+ * pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
+ channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_jingle.c, channels/chan_dahdi.c,
+ funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
+ funcs/func_aes.c: Add the attribute "type" to each "<use>" for
+ menuselect. This matters only when autoconf fails to detect that
+ weak linking is supported. External optional dependencies will
+ become optional in both cases, as they are removed at compile
+ time when not detected. However, runtime-optional modules are
+ made mandatory when weak linking is not found. This change
+ affects only the external optional dependencies; previously, they
+ were incorrectly required when weak linking support was not
+ detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt
+ by tilghman (License #5003) Tested by: iasgoscouk
+
+2011-07-05 17:22 +0000 [r326291] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Used auth=
+ parameter freed during "sip reload" causes crash. If you use the
+ auth= parameter and do a "sip reload" while there is an ongoing
+ call. The peer->auth data points to free'd memory. The patch does
+ several things: 1) Puts the authentication list into an ao2
+ object for reference counting to fix the reported crash during a
+ SIP reload. 2) Converts the authentication list from open coding
+ to AST list macros. 3) Adds display of the global authentication
+ list in "sip show settings". (closes issue ASTERISK-17939)
+ Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch
+ (license #5621) patch uploaded by rmudgett Review:
+ https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526
+
+2011-07-05 13:23 +0000 [r326209] Matthew Jordan <mjordan at digium.com>
+
+ * main/file.c: Updated filestream destructor to block until move is
+ complete when cache is used When a cache directory is used, the
+ process is forked and a mv command is executed to move the
+ temporary file to the permanent location. This caused issues with
+ voicemail, where a race condition occurred when the parent
+ expected the file to be in the permanent location prior to the mv
+ command completing. The parent process is now blocked until the
+ mv command completes. (closes issue ASTERISK-17724) Reported by:
+ Adiren P. Tested by: mjordan
+
+2011-07-01 21:07 +0000 [r326144] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Better way to get chan and pvt lock for
+ issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431
+ deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). *
+ Lock the channels in the defined order and avoid the need for a
+ deadlock avoidance loop. * Lock the channel before getting the
[... 32301 lines stripped ...]
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