[svn-commits] lmadsen: tag 1.6.2.12-rc1 r283280 - /tags/1.6.2.12-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Mon Aug 23 13:36:19 CDT 2010


Author: lmadsen
Date: Mon Aug 23 13:36:15 2010
New Revision: 283280

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=283280
Log:
Importing files for 1.6.2.12-rc1 release.

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    tags/1.6.2.12-rc1/.version   (with props)
    tags/1.6.2.12-rc1/ChangeLog   (with props)

Added: tags/1.6.2.12-rc1/.lastclean
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Added: tags/1.6.2.12-rc1/ChangeLog
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--- tags/1.6.2.12-rc1/ChangeLog (added)
+++ tags/1.6.2.12-rc1/ChangeLog Mon Aug 23 13:36:15 2010
@@ -1,0 +1,26846 @@
+2010-08-23  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.12-rc1 Released.
+
+2010-08-20 16:48 +0000 [r283049-283124]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
+	  (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
+	  https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
+	  | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
+	  line Reference correct struct member for unlikely event
+	  PRI_EVENT_CONFIG_ERR. .......... ................
+
+	* channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20
+	  Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending
+	  ALERTING is a protocol error The PRI layer in chan_dadhi will
+	  check if a PROGRESS message has already been sent, and not allow
+	  sending another (although that is technically allowed by the Q931
+	  spec), however it does not protect against sending an ALERTING
+	  and then sending a PROGRESS message, which is a violation of the
+	  specification. Most switches don't seem to care too deeply about
+	  this, but some do, and will disconnect the call when receiving
+	  this invalid sequence. Protocol specification reference:
+	  T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+	  protocol control (network side) point-point (sheet 3 of 8)"
+	  (closes issue #17874) Reported by: nic_bellamy Patches:
+	  asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+	  nic bellamy (license 299)
+	  asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+	  by nic bellamy (license 299)
+	  asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+	  by nic bellamy (license 299) ........
+
+2010-08-19 21:05 +0000 [r282890-282894]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
+	  | 11 lines tos_sip option was not being set correctly When
+	  tos_sip is used, the tos of the sip socket is only set correctly
+	  if the socket binding changes on a reload. If the binding stays
+	  the same but the TOS changes, the new tos value would not take
+	  into effect. This patch fixes that. (closes issue #17712)
+	  Reported by: nickb ........
+
+	* channels/chan_sip.c: fixes sip peer memory leaks in the
+	  peer_by_ip table (issue #17798)
+
+2010-08-19 19:44 +0000 [r282859]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Merged revisions 277944 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
+	  2010) | 16 lines Regression with T.38 negotiation Prior to
+	  1.4.26.3 T.38 negotiation worked properly, in the case of the
+	  reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+	  Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+	  by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+	  samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
+
+2010-08-19 02:14 +0000 [r282730]  Terry Wilson <twilson at digium.com>
+
+	* configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
+	  Aug 2010) | 2 lines Add some documentation about codec
+	  negotiation to sip.conf ........
+
+2010-08-18 14:28 +0000 [r282668]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: fixes crash with notifycid (closes issue
+	  #17868) Reported by: francesco_r Patches: issue_17868.diff
+	  uploaded by dvossel (license 671) Tested by: francesco_r
+
+2010-08-18 07:43 +0000 [r282607]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_dahdi.c: Don't warn on callerid when completely
+	  text, instead of numeric with localdialplan prefixes. (closes
+	  issue #16770) Reported by: jamicque Patches:
+	  20100413__issue16770.diff.txt uploaded by tilghman (license 14)
+	  20100811__issue16770.diff.txt uploaded by tilghman (license 14)
+	  Tested by: jamicque
+
+2010-08-17 21:35 +0000 [r282576]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: fixes no default transport for temp peer
+	  creation in chan_sip (closes issue #17829) Reported by: falves11
+	  Patches: issue_17829.rev1.txt uploaded by russell (license 2)
+	  issue_17829.diff uploaded by dvossel (license 671) Tested by:
+	  falves11
+
+2010-08-16 18:00 +0000 [r282469]  Leif Madsen <lmadsen at digium.com>
+
+	* doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information
+	  about creating sounds files using the sounds tools publically
+	  available so that others can create their own sounds prompts
+	  using the same tools we use to generate sounds releases. This
+	  allows people creating their own prompts to sound consistent with
+	  the prompts available from the open source project. SWP-595
+
+2010-08-16 17:32 +0000 [r282467]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c, /: Merged revisions 282430 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
+	  | 16 lines Send a SRCCHANGE indication when we masquerade
+	  Masquerading a channel means that the src of the audio is
+	  potentially changing, so send a SRCCHANGE so that RTP-based media
+	  streams can get a new SSRC generated to reflect the change.
+	  Original patch by addix (along with lots of testing--thanks!).
+	  (closes issue #17007) Reported by: addix Patches:
+	  1001-reset-SSRC-original-channel.diff uploaded by addix (license
+	  1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+	  addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+	  ........
+
+2010-08-13 18:54 +0000 [r282235]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: only do magic pickup when notifycid is
+	  enabled A new way of doing BLF pickup was introduced into 1.6.2.
+	  This feature adds a call-id value into the XML of a SIP_NOTIFY
+	  message sent to alert a subscriber that a device is ringing. This
+	  option should only be enabled when the new 'notifycid' option is
+	  set... but this was not the case. Instead the call-id value was
+	  included for every RINGING Notify message, which caused a
+	  regression for people who used other methods for call pickup.
+	  (closes issue #17633) Reported by: urosh Patches: chan_sip.txt
+	  uploaded by urosh (license ) blf_cid_issue.diff uploaded by
+	  dvossel (license 671) Tested by: dvossel, urosh, okrief,
+	  alecdavis
+
+2010-08-12 22:50 +0000 [r282130]  Jason Parker <jparker at digium.com>
+
+	* pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) |
+	  1 line Register CLI commands before parsing config, in case there
+	  is a config error. ........
+
+2010-08-12 03:01 +0000 [r281912]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, /: Merged revisions 281911 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
+	  | 20 lines Ensure SSRC is changed when media source is changed to
+	  resolve audio delay. This change causes the SSRC to change right
+	  before the channels are bridged, which is what used to happen. It
+	  seems that fixes were made to attempt limiting SSRC changes,
+	  targeted mainly at sending DTMF. DTMF is not affecting the SSRC
+	  with this change. There are two other control frames sent in
+	  ast_channel_bridge that probably should also be changed to
+	  AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
+	  up to the discretion of resolving issue #17007. For reference -
+	  old review implementing new control frame SRCCHANGE:
+	  https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+	  Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+	  (license 325) Tested by: sdolloff ........
+
+2010-08-11 21:09 +0000 [r281763-281873]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/say.conf.sample, /: Merged revisions 281819 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11
+	  Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes
+	  issue #17836) Reported by: RoadKill Patches:
+	  say.conf.sample.patch.dk uploaded by RoadKill (license 933)
+	  ........
+
+	* configs/say.conf.sample, /: Merged revisions 281762 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11
+	  Aug 2010) | 6 lines Allow say.conf to handle large numbers ending
+	  with multiple zeros. (closes issue #17833) Reported by: RoadKill
+	  Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
+	  (license 933) ........
+
+2010-08-11 15:17 +0000 [r281722]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_readexten.c: Only set status TIMEOUT, if we have no
+	  digits. (closes issue #15188) Reported by: jcovert Patches:
+	  app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
+	  551)
+
+2010-08-10 18:04 +0000 [r281567-281574]  Russell Bryant <russell at digium.com>
+
+	* main/sched.c: Don't move the time threshold for running scheduled
+	  events on every iteration. Instead, only calculate the time
+	  threshold each time ast_sched_runq() is called. (closes issue
+	  #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
+	  schmidts (license 1077)
+
+	* apps/app_dial.c, /: Merged revisions 281566 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+	  | 8 lines Reset visible indication after answer. (closes issue
+	  #17641) Reported by: klaus3000 Patches:
+	  ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+	  klaus3000 (license 65) Tested by: schmidts ........
+
+2010-08-09 20:46 +0000 [r281430]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: fixes SIP peers memory leak We zeroed out
+	  the peer's addr before it was removed from the peers_by_ip
+	  container. This made it impossible to be removed from the
+	  container as the addr is the key used by the container to find
+	  the peer. (closes issue #17774) Reported by: kkm Patches:
+	  017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
+	  017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
+
+2010-08-09 20:07 +0000 [r281391]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_local.c, /: Merged revisions 281390 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09
+	  Aug 2010) | 13 lines Prevent loss of Caller ID information set on
+	  local channel after masquerade. Caller ID set on the channel
+	  before a masquerade occurs when using a local channel would cause
+	  the information to be lost. The problem was that the information
+	  was set on a channel destined to be hung up. The somewhat
+	  confusing fix is to detect if any Caller ID has been set on the
+	  channel and if so preswap the Caller ID data so that basically
+	  the masquerade puts the data back. (closes issue #17138) Reported
+	  by: kobaz Review: https://reviewboard.asterisk.org/r/847/
+	  ........
+
+2010-08-05 13:11 +0000 [r281051]  Russell Bryant <russell at digium.com>
+
+	* main/cdr.c: Cleanup default option value handling for cdr.conf
+	  [general]. The default values would differ depending on whether
+	  or not cdr.conf exists. That is no longer the case. Apply a
+	  default value to the unanswered option. Define all default values
+	  as named constants.
+
+2010-08-05 07:40 +0000 [r280983]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982
+	  via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
+	  | 8 lines Change context lock back to a mutex, because
+	  functionality depends upon the lock being recursive. (closes
+	  issue #17643) Reported by: zerohalo Patches:
+	  20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+	  Tested by: zerohalo ........
+
+2010-08-03 20:52 +0000 [r280671-280812]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions
+	  280811 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010)
+	  | 9 lines Prevent DAHDI channels from overriding the callerid,
+	  once it's been set by the user. (closes issue #16661) Reported
+	  by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by
+	  tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt
+	  uploaded by tilghman (license 14) Tested by: jstapleton ........
+
+	* doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document
+	  -B and -W flags and regenerate manpage from sgml
+
+	* apps/app_voicemail.c: Allow the pipe, but also allow the comma
+
+2010-08-02 21:14 +0000 [r280669]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_sip.c: Change SIP NOTIFY requests to expect a
+	  response so authentication will work. This changes the request to
+	  be sent with the transmit type XMIT_RELIABLE so that sip_ack
+	  doesn't return false and cause the 401 to be ignored in cases
+	  where authentication is required. (closes issue #14255) Reported
+	  by: zktech
+
+2010-07-29 21:07 +0000 [r280556]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_config_curl.c: Off-by-one error (closes issue #17590)
+	  Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded
+	  by tilghman (license 14)
+
+2010-07-29 20:42 +0000 [r280449-280551]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: fixes wrong SRV query for TLS connection
+	  (closes issue #17612) Reported by: marcelloceschia Patches:
+	  chan-sip_srvQuery.patch uploaded by marcelloceschia (license
+	  1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
+	  chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
+	  (license 1079) Tested by: marcelloceschia, st, pabelanger
+
+	* main/channel.c, /: Merged revisions 280448 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
+	  | 12 lines fixes issue with translator frame not getting freed A
+	  translator frame even if it local storage so the translation path
+	  can be freed. This issue prevented g729 licenses from being freed
+	  up. (closes issue #17630) Reported by: manvirr Patches:
+	  encoder_fix.diff uploaded by dvossel (license 671) Tested by:
+	  manvirr, dvossel ........
+
+2010-07-29 16:01 +0000 [r280345]  Jean Galarneau <jgalarneau at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
+	  2 lines Fix a dsp structure leak occuring when a local channel is
+	  put into a meetme conference, then masquaraded away. ABE-2422
+	  ........
+
+2010-07-29 13:45 +0000 [r280306]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_local.c: Implement support for
+	  ast_channel_queryoption on local channels. Currently only
+	  AST_OPTION_T38_STATE is supported. ABE-2229 Review:
+	  https://reviewboard.asterisk.org/r/813/
+
+2010-07-28 20:02 +0000 [r280231]  Jason Parker <jparker at digium.com>
+
+	* sounds/Makefile: Work around some silly behavior on BSD. A
+	  non-zero exit from a subshell should make the build fail. (closes
+	  issue #17621)
+
+2010-07-28 19:57 +0000 [r280229]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Add missing enum value "unknown" to the
+	  SS7 called_nai and calling_nai config options.
+
+2010-07-28 19:54 +0000 [r280193-280227]  Jason Parker <jparker at digium.com>
+
+	* build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is
+	  no util on the system.
+
+	* sounds/Makefile: Remove unnecessary subshells. Attempt to make
+	  checksumming work. Also improves readability. (issue #17621)
+	  Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
+
+2010-07-28 16:51 +0000 [r280160]  Sean Bright <sean at malleable.com>
+
+	* apps/app_queue.c: Plug a reference leak in app_queue when adding
+	  members dynamically. (closes issue #17738) Reported by:
+	  bobwienholt Patches: issue17738.patch uploaded by bobwienholt
+	  (license 950) Tested by: bobwienholt, seanbright
+
+2010-07-28 13:51 +0000 [r280089]  Leif Madsen <lmadsen at digium.com>
+
+	* contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
+	  Jul 2010) | 1 line Update help text to be less confusing.
+	  ........
+
+2010-07-27 20:54 +0000 [r279946]  David Vossel <dvossel at digium.com>
+
+	* main/audiohook.c, main/channel.c, /,
+	  include/asterisk/audiohook.h: Merged revisions 279945 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
+	  | 19 lines remove empty audiohook write list on channel If a
+	  channel has an audiohook write list created on it, that list
+	  stays on the channel until the channel is destroyed. There is no
+	  reason to keep that list on the channel if it becomes empty. If
+	  it is empty that just means we are doing needless translating for
+	  every ast_read and ast_write. This patch removes the audiohook
+	  list from the channel once it is detected to be empty on either a
+	  read or write. If a audiohook is added back to the channel after
+	  this list is destroyed, the list just gets recreated as if it
+	  never existed to begin with. (closes issue #17630) Reported by:
+	  manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
+
+2010-07-27 17:54 +0000 [r279849-279883]  Jason Parker <jparker at digium.com>
+
+	* makeopts.in, configure, configure.ac: Add SHA1SUM to configure,
+	  since we require it for sounds/
+
+	* sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they
+	  aren't used anymore.
+
+	* sounds/Makefile: Simply sounds/Makefile some more.
+
+2010-07-27 15:13 +0000 [r279784]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static
+	  option in sip.conf. We were attempting to create a contactdeny
+	  rule based on the peer's IP address before the peer's IP address
+	  had been set. By moving the processing further down in the
+	  function, we can ensure stuff works as we expect for it to.
+	  (closes issue #17717) Reported by: mmichelson Patches:
+	  17717.patch uploaded by mmichelson (license 60) Tested by:
+	  DennisD
+
+2010-07-26 22:59 +0000 [r279657]  Jason Parker <jparker at digium.com>
+
+	* sounds/Makefile (added), sounds/Makefile.380 (removed),
+	  configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+	  (removed), configure.ac: Really fix sounds Makefile (and make it
+	  readableish). There was a rather large syntax error that should
+	  have caused ALL versions of GNU make to fail. I don't know how it
+	  worked.
+
+2010-07-26 21:18 +0000 [r279609]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, configure.ac: Dunno why this worked on my machine, but
+	  it works better this way.
+
+2010-07-26 20:25 +0000 [r279597]  Gavin Henry <ghenry at suretecsystems.com>
+
+	* res/res_config_ldap.c: Apply all patches in:
+	  https://issues.asterisk.org/view.php?id=13573 (closes issue
+	  #13573) Reported by: navkumar Patches:
+	  res_config_ldap-category.diff uploaded by navkumar (license 580)
+	  res_config_ldap.patch uploaded by bencer (license 961)
+	  res_config_ldap uploaded by bencer (license 961) Tested by:
+	  suretec
+
+2010-07-26 19:15 +0000 [r279561]  Tilghman Lesher <tlesher at digium.com>
+
+	* sounds/Makefile (removed), configure, sounds/Makefile.380
+	  (added), sounds/Makefile.381 (added), configure.ac: Use a special
+	  Makefile for noobs who still have GNU Make 3.80. (Closes issue
+	  #17716) Reported by: farisraouf
+
+2010-07-26 15:41 +0000 [r279501]  Sean Bright <sean at malleable.com>
+
+	* autoconf/ast_ext_lib.m4: Expand the correct value within
+	  AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
+
+2010-07-24 23:58 +0000 [r279347]  Bradley Latus <brad.latus at gmail.com>
+
+	* doc/asterisk.8: Minor update to man page
+
+2010-07-23 22:11 +0000 [r279207]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
+	  | 7 lines SIP promiscuous redirect could fail to dial the
+	  redirect. The ast_channel was created with one variable to
+	  ast_request() but the call to ast_call() that initiates the
+	  outgoing call was using a different variable. The two variables
+	  are not equivalent if the call_forward string included a channel
+	  technology specifier. e.g., SIP/200 ........
+
+2010-07-23 18:29 +0000 [r279112]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk
+	  to 1.6.2.
+
+2010-07-23 18:22 +0000 [r279072-279088]  Russell Bryant <russell at digium.com>
+
+	* /: remove old properties
+
+	* /: Add branch-1.4-merged and branch-1.4-blocked properties to
+	  1.6.2 branch.
+
+2010-07-23 17:06 +0000 [r278983-278986]  Tilghman Lesher <tlesher at digium.com>
+
+	* autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
+	  revisions 278985 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010)
+	  | 12 lines Merged revisions 278984 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
+	  | 5 lines Establish a maximum version for openh323 (i.e. not
+	  opal), because chan_h323 will fail to load, even if it links.
+	  (issue #17679) Reported by: am ........ ................
+
+	* main/asterisk.c, /: Merged revisions 278982 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010)
+	  | 15 lines Merged revisions 278981 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
+	  | 8 lines Avoid race with consolethread on shutdown (on parallel
+	  processors). (closes issue #17080) Reported by: sybasesql
+	  Patches: 20100721__issue17080.diff.txt uploaded by tilghman
+	  (license 14) Tested by: sybasesql ........ ................
+
+2010-07-23 15:23 +0000 [r278934]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* channels/chan_dahdi.c: Two more typos to cancell.
+
+2010-07-22 19:52 +0000 [r278709]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/xmldoc.c, /: Merged revisions 278708 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 |
+	  jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
+	  Add method for finding XML doc files for systems that don't
+	  support GLOB_BRACE. In particular, Solaris and perhaps others do
+	  not support the above mentioned GNU extension. In this case the
+	  paths are simply expanded without the braces and the calls to
+	  glob are made separately. Note: I could not explain memory
+	  allocation failures that were being reported from within libxml
+	  itself when making calls to glob without using GLOB_NOCHECK. This
+	  is the only reason why that flag is being used. (closes issue
+	  #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded
+	  by snuffy (license 35), modified by me ........
+
+2010-07-22 19:32 +0000 [r278703]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: DNID does not get cleard on a new call
+	  when using immediate=yes with ISDN signaling. When you are using
+	  chan_dahdi ISDN signaling with immediate=yes and a call comes in
+	  without a DNID then you get the DNID of a previous call.
+	  Chan_dahdi does not touch the DNID field on a new call if it does
+	  not have a DNID. Made always copy the DNID from the new call. The
+	  patches backport the relevant changes from trunk -r210387.
+	  (closes issue #17568) Reported by: wuwu Patches:
+	  issue17568_v1.4.patch uploaded by rmudgett (license 664)
+	  issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
+
+2010-08-10  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.11 Released.
+
+2010-07-26  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.11-rc2 Released.
+
+2010-07-26  Leif Madsen <lmadsen at digium.com>
+
+	* qwell, asterisk, branch-1.6.2, r279657 ***
+	  Really fix sounds Makefile (and make it readableish).
+	  There was a rather large syntax error that should have
+	  caused ALL versions of GNU make to fail.
+	  I don't know how it worked.
+
+	  (Closes issue #17716)
+
+2010-07-22  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.11-rc1 Released.
+
+2010-07-22 15:00 +0000 [r278621]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c, /: Merged revisions 278620 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul
+	  2010) | 19 lines Merged revisions 278618 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
+	  2010) | 13 lines Allow PLC to function properly when channels use
+	  SLIN for audio. If a channel involved in a bridge was using SLIN
+	  audio, then translation paths were not guaranteed to be set up
+	  properly since in all likelihood the number of translation steps
+	  was only 1. This patch enforces the transcode_via_slin behavior
+	  if transcode_via_slin or generic_plc is enabled and one of the
+	  formats to make compatible is SLIN. AST-352 ........
+	  ................
+
+2010-07-21 18:22 +0000 [r278524]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO
+	  channels This fixes some cases of no outgoing calls on FXO before
+	  an incoming call. Remove an unnecessary testing of an "off-hook"
+	  bit from DAHDI for FXO (KS/GS) channels.In some cases the bit
+	  would not be initialized properly before the first inbound call
+	  and thus prevent an outgoing call. If those tests are actually
+	  required by anybody, they should define DAHDI_CHECK_HOOKSTATE in
+	  channels/sig_analog.c . (closes issue #14577) Reported by: jkroon
+	  Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd
+	  (license 610) Tested by: frawd Review:
+	  https://reviewboard.asterisk.org/r/699/
+
+2010-07-21 16:20 +0000 [r278479]  Russell Bryant <russell at digium.com>
+
+	* /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010)
+	  | 41 lines Use poll() instead of select() in res_timing_pthread
+	  to avoid stack corruption. This code did not properly check
+	  FD_SETSIZE to ensure that it did not try to select() on fds that
+	  were too large. Switching to poll() removes the limitation on the
+	  maximum fd value. (closes issue #15915) Reported by: keiron
+	  (closes issue #17187) Reported by: Eddie Edwards (closes issue
+	  #16494) Reported by: Hubguru (closes issue #15731) Reported by:
+	  flop (closes issue #12917) Reported by: falves11 (closes issue
+	  #14920) Reported by: vrban (closes issue #17199) Reported by:
+	  aleksey2000 (closes issue #15406) Reported by: kowalma (closes
+	  issue #17438) Reported by: dcabot (closes issue #17325) Reported
+	  by: glwgoes (closes issue #17118) Reported by: erikje possibly
+	  other issues, too ... ........
+
+2010-07-21 15:58 +0000 [r278025-278464]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 |
+	  tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11
+	  lines Ensure realtime conferences are treated the same as static
+	  conferences when trying to find an empty one. Also, parse the
+	  useropts properly, when retrieving from realtime, and add them to
+	  the existing flags. (closes issue #17502) Reported by: kenji
+	  Patches: 20100720__issue17502.diff.txt uploaded by tilghman
+	  (license 14) Tested by: kenji ........
+
+	* apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500
+	  (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010)
+	  | 7 lines Delete IMAP messages in reverse order, to ensure
+	  reordering after each expunge does not cause deletion of the
+	  wrong message. (closes issue #16350) Reported by: noahisaac
+	  Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+	  (license 14) ........ ................
+
+	* main/autoservice.c, /, main/features.c,
+	  include/asterisk/channel.h: Merged revisions 278272 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500
+	  (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010)
+	  | 4 lines Do not queue up DTMF frames while a call is on hold.
+	  (Fixes ABE-2110) ........ ................
+
+	* main/manager.c, /: Merged revisions 278024 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010)
+	  | 14 lines Merged revisions 278023 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
+	  | 7 lines Off-by-one error (closes issue #16506) Reported by:
+	  nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+	  tilghman (license 14) ........ ................
+
+2010-07-19 21:21 +0000 [r277966]  Jean Galarneau <jgalarneau at digium.com>
+
+	* /, main/features.c: Merged revisions 277945 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) |
+	  15 lines Merged revisions 277906 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
+	  7 lines Avoid trying to pickup a parked extension before the park
+	  operation is completed. A crash could occur if the extension is
+	  picked up while the parking extension is being announced. Testing
+	  pu->notquiteyet while searching for a parked extension resolves
+	  this crash. (ABE-2418) ........ ................
+
+2010-07-17 17:52 +0000 [r277774-277777]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_config_pgsql.c: Merge issues...
+
+	* /, autoconf/ast_func_fork.m4, configure,
+	  include/asterisk/autoconfig.h.in: Merged revisions 277775 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500
+	  (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
+	  | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
+	  working fork()... it's only uclinux that does not. (closes issue
+	  #17616) Reported by: pprindeville ........ ................
+
+	* res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+	  revisions 277773 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010)
+	  | 15 lines Merged revisions 277568 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010)
+	  | 8 lines Since we split values at the semicolon, we should store
+	  values with a semicolon as an encoded value. (closes issue
+	  #17369) Reported by: gkservice Patches:
+	  20100625__issue17369.diff.txt uploaded by tilghman (license 14)
+	  Tested by: tilghman ........ ................
+
+2010-07-16 23:37 +0000 [r277666]  Tim Ringenbach <tim.ringenbach at gmail.com>
+
+	* /, main/features.c: Merged revisions 277657 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul
+	  2010) | 16 lines Merged revisions 277625 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
+	  2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
+	  attended transfer. ast_bridge_call() clears
+	  AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+	  ast_bridge_call() is called for a second bridge on the same
+	  channel, and it clears that flag, which still needs to get set
+	  for when the original ast_bridge_call() gets control back and
+	  checks it. Review: https://reviewboard.asterisk.org/r/741
+	  ........ ................
+
+2010-07-16 21:31 +0000 [r277563]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul
+	  2010) | 11 lines Merged revisions 277497 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
+	  2010) | 4 lines Default to no udptl error correction so that
+	  error correction will be disabled in the event that the remote
+	  end indicates that they do not support the error correction mode
+	  we requested. FAX-128 ........ ................
+
+2010-07-16 21:16 +0000 [r277489]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 277488 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 |
+	  jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
+	  Fix reporting estimated queue hold time. Just say the number of
+	  seconds (after minutes) rather than doing some incorrect
+	  calculation with respect to minutes. (closes issue #17498)
+	  Reported by: corruptor Patches: holdesecs_bug.diff uploaded by
+	  corruptor (license 253) ........
+
+2010-07-16 20:35 +0000 [r277485]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500
+	  (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010)
+	  | 15 lines priexclusive in chan_dahdi.conf ignored when reloading
+	  dahdi module During a reload, the priexclusive and outsignalling
+	  parameters are not read in from the config file as intended.
+	  Unfortunately, they get set to defaults as a result. This patch
+	  makes sure that they do not get set to defaults during a reload.
+	  (closes issue #17441) Reported by: mtryfoss Patches:
+	  issue17441_v1.4.patch uploaded by rmudgett (license 664)
+	  issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+	  issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+	  by: rmudgett ........ ................
+
+2010-07-16 20:30 +0000 [r277478]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
+	  (added), /: Merged revisions 277452 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 |
+	  tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
+	  Add documentation for MOH realtime fields ........
+
+2010-07-16 19:24 +0000 [r277377]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 277366 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 |
+	  jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
+	  Add missing handling for ringing state for use with queue empty
+	  options. (closes issue #17471) Reported by: jazzy Patches:

[... 26109 lines stripped ...]



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