[asterisk-commits] bebuild: tag 1.8.6.0-rc1 r331374 - /tags/1.8.6.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Aug 10 11:09:54 CDT 2011


Author: bebuild
Date: Wed Aug 10 11:09:49 2011
New Revision: 331374

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=331374
Log:
Importing files for 1.8.6.0-rc1 release.

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    tags/1.8.6.0-rc1/ChangeLog   (with props)

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--- tags/1.8.6.0-rc1/ChangeLog (added)
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+2011-08-10  Asterisk 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
+	  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>
+

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