[asterisk-commits] lmadsen: tag 1.8.0-rc5 r292275 - /tags/1.8.0-rc5/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Oct 18 17:27:22 CDT 2010


Author: lmadsen
Date: Mon Oct 18 17:27:19 2010
New Revision: 292275

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=292275
Log:
Importing files for 1.8.0-rc5 release.

Added:
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    tags/1.8.0-rc5/.version   (with props)
    tags/1.8.0-rc5/ChangeLog   (with props)

Added: tags/1.8.0-rc5/.lastclean
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--- tags/1.8.0-rc5/ChangeLog (added)
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@@ -1,0 +1,25500 @@
+2010-10-18  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-rc5 Released.
+
+2010-10-18 22:02 +0000 [r292230]  Leif Madsen <lmadsen at digium.com>
+
+	* sounds/Makefile, /: Merged revisions 292229 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010)
+	  | 3 lines Fix typo in the sounds/Makefile. (Issue #17426)
+	  ........
+
+2010-10-18 21:55 +0000 [r292227]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 292226 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500
+	  (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010)
+	  | 11 lines Fix improper operator key acceptance and clean up temp
+	  recording files. This is a fix for when pressing the operator key
+	  after recording an unavailable, busy, name, or temporary message
+	  in mailbox options. The operator key should not be accepted here,
+	  but should be allowed during the message recording. If the
+	  operator key is pressed during ensure the file is saved or
+	  deleted as apporopriate. Also, ensure removal of temporary
+	  recorded files after an early hang up or when message acceptance
+	  confirmation times out. ABE-2518 ........ ................
+
+2010-10-18 21:51 +0000 [r292225]  Leif Madsen <lmadsen at digium.com>
+
+	* sounds/sounds.xml, sounds/Makefile, /: Merged revisions 292224
+	  via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500
+	  (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
+	  | 9 lines Add support for the new English (Australian Accent)
+	  sound files. (closes issue #17426) Reported by: camsown Patches:
+	  core-sounds-en_AU.txt uploaded by camsown (license 1050)
+	  add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
+	  by: camsown, lmadsen, jtodd, qwell ........ ................
+
+2010-10-18 19:50 +0000 [r292188]  Russell Bryant <russell at digium.com>
+
+	* main/netsock2.c: Resolve some compiler errors in
+	  ast_sockaddr_is_any(). These errors came up once this function
+	  was used from within netsock2.c. The errors were like the
+	  following: netsock2.c:393: error: dereferencing pointer
+	  ‘({anonymous})’ does break strict-aliasing rules The usage of a
+	  union here avoids this problem.
+
+2010-10-18 19:16 +0000 [r292155]  David Vossel <dvossel at digium.com>
+
+	* main/netsock2.c: Fixes build error for systems not supporting
+	  IPV6_TCLASS.
+
+2010-10-18 17:15 +0000 [r292122]  Matthew Nicholson <mnicholson at digium.com>
+
+	* addons/chan_mobile.c: Fix the cmgr parser. (closes issue 0018152)
+	  Reported by: menschentier
+
+2010-10-18  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-rc4 Released
+
+2010-10-18 16:02 +0000 [r292085]  David Vossel <dvossel at digium.com>
+
+	* main/netsock2.c: Fixes qos settings for sockets bound to any IPv6
+	  or IPv4 address. (closes issue #18099) Reported by: jamesnet
+	  Patches: issues_18099_v3.diff uploaded by dvossel (license 671
+
+2010-10-18 15:32 +0000 [r292083]  Jeff Peeler <jpeeler at digium.com>
+
+	* pbx/pbx_spool.c: Disable use of inotify for call file handling as
+	  it is not working properly. (related to #18089)
+
+2010-10-16 10:47 +0000 [r292050]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* res/res_musiconhold.c, /, configs/musiconhold.conf.sample: Merged
+	  revisions 292049 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) |
+	  15 lines Base directory for MOH should be ASTDATADIR If the
+	  directive 'directory' is relative, make it relative to the
+	  datadir, rather than to the varlibdir. In the sample
+	  configuration it is relative ('moh'). This has no effect unless
+	  you have actively set the datadir explicitly (at build time or at
+	  run time). (closes issue #16906) Patches: moh_datadir uploaded by
+	  tzafrir (license 46) Review:
+	  https://reviewboard.asterisk.org/r/974/ ........
+
+2010-10-15 21:40 +0000 [r292016]  Terry Wilson <twilson at digium.com>
+
+	* res/res_srtp.c: Ref/unref res_srtp when we create/destroy a
+	  session This avoids unhappy crashing when we try to 'core stop
+	  gracefully' and res_srtp tries to unload before chan_sip does.
+	  Thanks, Russell! (closes issue #18085) Reported by: st
+
+2010-10-15 20:12 +0000 [r291942]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: Fixes peer's host port information being
+	  lost on sip reload. (closes issue #18135) Reported by: lmadsen
+	  Patches: crazy_ports_v2.diff uploaded by dvossel (license 671)
+	  Tested by: lmadsen
+
+2010-10-15 19:50 +0000 [r291940]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* configs/gtalk.conf.sample, /: Merged revisions 291939 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400
+	  (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri,
+	  15 Oct 2010) | 2 lines Clean up formatting. ........
+	  ................
+
+2010-10-15 16:39 +0000 [r291905]  Terry Wilson <twilson at digium.com>
+
+	* res/res_jabber.c, /: Merged revisions 291904 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010)
+	  | 7 lines Don't crash or deadlock on module unload We can't hold
+	  the lock while pthread_join is called since aji_log_hook will
+	  attempt to lock from the other therad. We reorder the
+	  pthread_join and ast_aji_disconnect so that we don't do an
+	  SSL_read() while SSL_shutdown is running, causing a crash.
+	  ........
+
+2010-10-14 22:09 +0000 [r291827-291829]  David Vossel <dvossel at digium.com>
+
+	* main/netsock2.c: Set TCLASS field of IPv6 header when sip qos
+	  options are set. (closes issue #18099) Reported by: jamesnet
+	  Patches: issues_18099_v2.diff uploaded by dvossel (license 671)
+	  Tested by: dvossel, jamesnet
+
+	* channels/chan_gtalk.c: Safer xml parsing, treat all clients the
+	  same, and better local candidate selection. The gtalk channel
+	  driver was doing several unsafe operations in regards to how it
+	  parsed incoming XML messages. I have cleaned that code up so it
+	  should be much safer now. We now treat all clients types the
+	  same. We have no reason to distinguish between GMAIL and GOOGLE
+	  VOICE clients anymore because they all work the same way. I also
+	  modified how the local ip is found. If no bindaddress is provided
+	  in the config file, we attempt to determine the local ip we would
+	  use to connect to google.com. If that fails, then we fall back to
+	  the ast_find_ourip() function as a last resort. Using the new
+	  method makes it much less likely that we would ever advertise a
+	  local RTP candidate as a loopback address.
+
+2010-10-14 18:45 +0000 [r291791]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/stdtime/localtime.c: Add missing ifdefs for test framework
+	  and new locale code. (closes issue #18137) Reported by: ovi
+	  Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes
+	  (license 717) 18137_localelist_warning.patch uploaded by wdoekes
+	  (license 717) Tested by: ovi
+
+2010-10-14 15:15 +0000 [r291758]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_gtalk.c, channels/chan_jingle.c,
+	  include/asterisk/acl.h, channels/chan_sip.c,
+	  channels/chan_h323.c, main/acl.c: Add the ability for
+	  ast_find_ourip to return IPv4, IPv6 or both. While testing
+	  chan_gtalk I noticed jabber was using my IPv6 address and not
+	  IPv4. When using bindaddr=0.0.0.0 it is possible for
+	  ast_find_ourip() to return both IPv6 and IPv4 results. Adding a
+	  family parameter gives you the ablility to choose. Since
+	  jabber/gtalk/h323 do not support IPv6, we should only return IPv4
+	  results. Review: https://reviewboard.asterisk.org/r/973/
+
+2010-10-14 12:08 +0000 [r291725]  Russell Bryant <russell at digium.com>
+
+	* doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/
+
+2010-10-13 23:45 +0000 [r291656]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c, /,
+	  channels/sig_analog.h: Merged revisions 291655 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500
+	  (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010)
+	  | 20 lines Deadlock between dahdi_exception() and
+	  dahdi_indicate(). There is a deadlock between dahdi_exception()
+	  and dahdi_indicate() for analog ports. The call-waiting and
+	  three-way-calling feature can experience deadlock if these
+	  features are trying to do something and an event from the bridged
+	  channel happens at the same time. Deadlock avoidance code added
+	  to obtain necessary channel locks before attemting an operation
+	  with call-waiting and three-way-calling. (closes issue #16847)
+	  Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+	  uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+	  uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+	  uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+	  Review: https://reviewboard.asterisk.org/r/971/ ........
+	  ................
+
+2010-10-13 23:01 +0000 [r291581]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c, /: Merged revisions 291580 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r291580 | twilson | 2010-10-13 15:58:43 -0700
+	  (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
+	  | 21 lines Don't ignore frames that have been queued when
+	  softhangup'd When an outgoing call is answered and hung up by the
+	  far end *very* quickly, we may not read any frames and therefor
+	  end up with a call that displays the wrong
+	  disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+	  immediately sets the _softhangup flag on the channel and then
+	  queues the HANGUP control frame, but __ast_read refuses to read
+	  any frames if ast_check_hangup() indicates that a hangup request
+	  has been made (which it will if _softhangup is set). So, we end
+	  up losing control frames. This change makes __ast_read continue
+	  to read frames even if a soft hangup has been requested. It
+	  queues a hangup frame to make sure that __ast_read() will still
+	  eventually return NULL. Much thanks to David Vossel for all of
+	  the reviews, discussion, and help! (closes issue #16946) Reported
+	  by: davidw Review: https://reviewboard.asterisk.org/r/740/
+	  ........ ................
+
+2010-10-13 22:46 +0000 [r291578]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_gtalk.c: More fixup for chan_gtalk. This patch
+	  makes the xml parsing safer.
+
+2010-10-13 22:24 +0000 [r291575]  Terry Wilson <twilson at digium.com>
+
+	* Makefile, static-http/mantest.html (added): Add a simple AMI
+	  client web page This patch uses the XML docs to parse all of the
+	  available AMI commands and allows you to enter the command name
+	  and be presented with a form with the available fields. You can
+	  then rapidly tab through the fields and submit the command and
+	  view the response. It is much faster/easier than having to use
+	  telnet for testing purposes.
+
+2010-10-13 20:21 +0000 [r291469-291541]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: The chan_dahdi faxdetect option only works
+	  for the first FAX call. The chan_dahdi faxdetect option only
+	  works for the first call. After that the option no longer works.
+	  The struct dahdi_pvt.callprogress member is the encoded user
+	  config setting for the callprogress and faxdetect config options.
+	  Changing this value alters the configuration for all following
+	  calls until the chan_dahdi.conf file is reloaded. * Fixed the
+	  chan_dahdi ast_channel_setoption callback to not change the users
+	  faxdetect config setting except for the current call. * Fixed the
+	  chan_dahdi ast_channel_queryoption callback to read the active
+	  DSP setting of the faxdetect option. * Made actually disable the
+	  active faxdetect DSP setting for the current call on the analog
+	  port. my_handle_dtmfup() is used for normal analog ports.
+	  dahdi_handle_dtmfup() is the legacy code and is no longer used
+	  unless in a radio mode. (closes issue #18116) Reported by:
+	  seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett
+	  (license 664) Review: https://reviewboard.asterisk.org/r/972/
+
+	* channels/chan_misdn.c: Merged revision 291504 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed,
+	  13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the
+	  ast_channel. Must get the ast_channel lock before proceeding with
+	  release_chan() and release_chan_early() to hold off ast_hangup()
+	  from destroying the ast_channel. Missed this change for -r291468.
+	  JIRA ABE-2598 JIRA SWP-2317 ..........
+
+	* channels/chan_misdn.c: Merge revision 291468 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed,
+	  13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN
+	  call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE
+	  --> RELEASE_COMPLETE * Add lock protection around channel list
+	  for find/add/delete operations. * Protect misdn_hangup() from
+	  release_chan() and vise versa using the release_lock. JIRA
+	  ABE-2598 JIRA SWP-2317 ..........
+
+2010-10-13 15:46 +0000 [r291394]  Russell Bryant <russell at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 291393 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r291393 | russell | 2010-10-13 10:29:21 -0500
+	  (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
+	  | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
+	  a frame. This fixes a crash due to a hangup race condition.
+	  ABE-2601 ........ ................
+
+2010-10-12 17:20 +0000 [r291284]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/phoneprov.conf.sample, /: Merged revisions 291280 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010)
+	  | 7 lines Add undocumented variables to phoneprov.conf.sample
+	  (closes issue #18107) Reported by: lathama Patches:
+	  phoneprov.conf.sample.diff uploaded by lathama (license 1028)
+	  ........
+
+2010-10-12 17:06 +0000 [r291265]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/acl.c: Merged revisions 291264 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500
+	  (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12
+	  Oct 2010) | 2 lines Oops, incorrect range (although unallocated
+	  at ARIN) ........ ................
+
+2010-10-12 16:08 +0000 [r291230]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/manager.conf.sample, /: Merged revisions 291229 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010)
+	  | 2 lines Add documention that mentions options are defined but
+	  not used. (Issue #18101) ........
+
+2010-10-12 15:58 +0000 [r291192-291227]  David Vossel <dvossel at digium.com>
+
+	* main/manager.c: Fixes manager.c crash. This issue was caused by
+	  improper use of the mansession lock and manession_session lock.
+	  These two structures are confusing to begin with so I'm not
+	  surprised this occurred. I fixed this by consistently making sure
+	  we use each of these locks only to protect the data in the
+	  corresponding structure. We had mismatched usage of these locks
+	  which resulted in no mutual exclusivity occurring at all. (closes
+	  issue #17994) Reported by: vrban Patches:
+	  mansession_locking_fix.diff uploaded by dvossel (license 671)
+	  Tested by: vrban
+
+	* CHANGES: Update CHANGES to reflect new gtalk.conf options.
+
+	* channels/chan_gtalk.c, include/asterisk/stun.h,
+	  configs/gtalk.conf.sample, res/res_stun_monitor.c: Gtalk
+	  enhancements and general code cleanup. This patch includes
+	  several chan_gtalk enhancements. Two new gtalk.conf options have
+	  been added, externip and stunadd. Setting externip allows us to
+	  manually specify what the external IP address is outside of a NAT
+	  environment. Setting the stunaddr option to a valid stun server
+	  allows for that external ip to be retrieved via a STUN server
+	  automatically. This external IP is then advertised during call
+	  setup as a possible candidate. I have also attempted to clean up
+	  chan_gtalk's code so it meets our coding guidelines. During this
+	  cleanup I noticed several things that need to be done in the code
+	  and made a TODO section at the top of the file.
+
+2010-10-11 18:51 +0000 [r291075-291113]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_sip.c: Move declaration closer to where now used.
+
+	* /, channels/chan_sip.c: Merged revisions 291110-291111 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500
+	  (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11
+	  Oct 2010) | 1 line Add missing unlock to an exception condition
+	  in reload_config(). ........ ................ r291111 | rmudgett
+	  | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit
+	  from handle_request_do() consistent. ................
+
+	* main/cli.c, /: Merged revisions 291073 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010)
+	  | 15 lines Fixed infinite loop in verbose/debug message output.
+	  Setting the module/filename specific message level and then
+	  changing it resulted in the linked list being looped on itself.
+	  Traversing this linked list is an infinite loop if what you are
+	  looking for is not in the list. Also plugged some CLI parsing
+	  holes in the associated CLI command: * Removing a nonexistent
+	  module from the list actually added it with a level of zero. *
+	  Setting the non-module specific level to zero is now equivalent
+	  to setting it to "off" as documented. ........
+
+2010-10-09 23:25 +0000 [r291038]  Tilghman Lesher <tlesher at digium.com>
+
+	* cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Add missing
+	  option to set calls to be logged in GMT/UTC.
+
+2010-10-09 15:00 +0000 [r291005-291037]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/oochannels.c: small correction for verbose
+	  print h.323 packets
+
+	* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+	  addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling
+	  options per user and peer. Added options for faststart/h.245
+	  tunneling per user/peer, properly handle these and global
+	  options, correction of handling fs/tunneling fields in signalling
+	  responses (issue #17972) Reported by: salecha Patches:
+	  fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
+	  Tested by: may213, salecha
+
+2010-10-08 20:44 +0000 [r290973]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_gtalk.c: Make outbound Google Voice calls. This
+	  patch allows for outbound Google Voice calls to be dialed from
+	  Asterisk using chan_gtalk. Below is an example dialstring. exten
+	  -> blah,1,Dial(Gtalk/asterisk/+15552225555 at voice.google.com,,) In
+	  this example, 'asterisk' is the jabber.conf profile configured to
+	  connect to your gmail account. In order to receive Google Voice
+	  calls make sure to enable 'allowguest=yes' in gtalk.conf.
+
+2010-10-08 15:49 +0000 [r290937-290938]  Erin Spiceland <erin at thespicelands.com>
+
+	* addons/res_config_mysql.c: Parentheses around assignment used as
+	  truth value, introduced in r290937.
+
+	* addons/res_config_mysql.c, addons/app_mysql.c,
+	  configs/res_config_mysql.conf.sample: Add option to
+	  res_config_mysql and app_mysql to specify a character set that
+	  MySQL should use. (closes issue 17948) Reported by qmax.
+
+2010-10-08 02:56 +0000 [r290864]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/asterisk.c, /: Merged revisions 290863 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500
+	  (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
+	  | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
+	  at control console. A recent change was made to avoid a race
+	  condition on shutdown which only called the end functions from
+	  the console thread. However, when pressing Ctrl-C the quit
+	  handler is called from the signal handler thread. (closes issue
+	  #17698) Reported by: jmls ........ ................
+
+2010-10-07 22:38 +0000 [r290828-290829]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_gtalk.c: Add Philippe Sultan to chan_gtalk author
+	  list. Philippe has made some notable contributions to the gtalk
+	  channel driver. His name deserves to be listed amoung the authors
+	  of that file. Thanks Philippe!
+
+	* channels/chan_gtalk.c: Outbound gtalk calls now work correctly.
+	  There was a problem with how the candidates were being built on
+	  an outbound call. This patch fixes that.
+
+2010-10-07 20:58 +0000 [r290752]  Jason Parker <jparker at digium.com>
+
+	* autoconf/ast_ext_lib.m4, /, configure,
+	  include/asterisk/autoconfig.h.in: Merged revisions 290751 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r290751 | qwell | 2010-10-07 15:57:14 -0500
+	  (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
+	  9 lines Allow PRI to build properly when using --with-pri. Use
+	  the directories found for the parent when using lib dependencies.
+	  (closes issue #17314) Reported by: tzafrir Patches:
+	  17314-withdeps.diff uploaded by qwell (license 4) ........
+	  ................
+
+2010-10-07  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-rc3 Released.
+
+2010-10-07 11:00 +0000 [r290713]  Russell Bryant <russell at digium.com>
+
+	* main/pbx.c, /: Merged revisions 290712 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
+	  | 4 lines Don't crash when Set() is called without a value.
+	  Review: https://reviewboard.asterisk.org/r/949/ ........
+
+2010-10-06 21:22 +0000 [r290648-290674]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_gtalk.c: Fixes commented out code to use #if 0
+	  instead. Thanks to rmudgett for catching this!
+
+	* channels/chan_gtalk.c: Fixes gtalk outbound DTMF to work
+	  properly. Outbound DTMF with gtalk needs to be done within the
+	  RTP stream. I discovered this after investigating a packet
+	  capture from the gmail client. Instead of performing jingle
+	  signaling DTMF, the gtalk servers expect all DTMF to arrive on
+	  the RTP stream using RFC2833 way of doing things. Chan_gtalk also
+	  had an issue with negotiating RTP payload type 106 for the
+	  telephony-event and then sending DTMF as payload 101. This has
+	  been resolved by always negotiating 101 as the payload type like
+	  we do everywhere else. With this patch, incoming google voice
+	  calls forwarded to Asterisk via gtalk work.
+
+2010-10-06 18:50 +0000 [r290614]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_dial.c: Merged revision 290613 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
+	  06 Oct 2010) | 5 lines Eliminate a redundant test for
+	  AST_CONTROL_REDIRECTING. Eliminate redundant test for
+	  AST_CONTROL_REDIRECTING that prevents running the redirecting
+	  interception macro if it is defined. ..........
+
+2010-10-06 13:49 +0000 [r290576]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/file.c: Merged revisions 290575 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
+	  | 8 lines Allow streaming audio from a pipe. (closes issue
+	  #18001) Reported by: jamicque Patches:
+	  20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+	  Tested by: jamicque ........
+
+2010-10-06 04:35 +0000 [r290542]  Terry Wilson <twilson at digium.com>
+
+	* res/res_rtp_asterisk.c: Don't try to send RTP when remote_address
+	  is null It is possible for ast_rtp_stop() to be called which will
+	  clear the remote address and cause the sendto to fail and spam
+	  warnings. Don't send in this case.
+
+2010-10-05 22:23 +0000 [r290479-290506]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Fixes uninitialized memory problem in 'iax2
+	  set debug peer' option.
+
+	* include/asterisk/jingle.h, channels/chan_gtalk.c,
+	  res/res_jabber.c, include/asterisk/jabber.h: Fixes chan_gtalk to
+	  work with gmail client This patch was written by Philippe Sultan
+	  (phsultan). Thanks for keeping this up to date!
+
+2010-10-05 20:23 +0000 [r290408]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_jabber.c, /: Merged revisions 290396 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
+	  (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+	  | 8 lines Fix a crash by ensuring that we don't alter memory
+	  after it's freed. (closes issue #17387) Reported by: jmls
+	  Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+	  (license 14) Tested by: jmls ........ ................
+
+2010-10-05 20:09 +0000 [r290376-290378]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Resolves dnsmgr memory corruption in
+	  chan_iax2. (closes issue #17902) Reported by: afried Patches:
+	  issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+	  afried, russell, dvossel Review:
+	  https://reviewboard.asterisk.org/r/965/
+
+	* /, apps/app_directed_pickup.c: Merged revisions 290375 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
+	  | 10 lines Fixes PickupChan() not working with full channel name.
+	  (closes issue #18011) Reported by: schern Patches:
+	  app_directed_pickup.c.2.patch uploaded by schern (license 995)
+	  app_directed_pickup.c.trunk.patch uploaded by schern (license
+	  995) Tested by: schern, dvossel ........
+
+2010-10-05 14:15 +0000 [r290066-290289]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, configure.ac: Restore run directory for OS X, as well
+	  as standardizing some other paths to Mac OS X.
+
+	* pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
+	  pbx/ael/ael-test/ref.ael-test19,
+	  pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
+	  pbx/ael/ael-test/ref.ael-vtest17, /,
+	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3:
+	  Merged revisions 290254 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
+	  | 11 lines Change new pattern matcher to regard dashes the same
+	  as the old pattern matcher -- as visual candy to be ignored. Also
+	  change the AEL parser to not generate dashes within extensions,
+	  as those dashes would be ignored. Update the AEL tests to match
+	  this behavior. (closes issue #17366) Reported by: murf Patches:
+	  20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+	  Tested by: tilghman ........
+
+	* /, configure, configure.ac: Merged revisions 290201 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
+	  (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+	  Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+	  ................
+
+	* /, configure, configure.ac: Merged revisions 290101 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
+	  (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+	  Oct 2010) | 2 lines Automatically re-run configure test for
+	  menuselect, when the relevant makeopts settings change. ........
+	  ................
+
+	* pbx/pbx_spool.c: Get notification only when file is closed, not
+	  when created. (closes issue #17924) Reported by: mkeuter Patches:
+	  asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
+	  Tested by: abelbeck
+
+2010-10-02 17:57 +0000 [r290026]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* contrib/scripts/get_mp3_source.sh: Allow users to pass additional
+	  arguments to the Subversion command that obtains the MP-3 source
+	  code. (reported on IRC by jmls)
+
+2010-10-02 08:56 +0000 [r289951]  Olle Johansson <oej at edvina.net>
+
+	* main/manager.c, /: Merged revisions 289950 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
+	  02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+	  lines Add documentation for undocumented option to AMI action
+	  originate ........ ................
+
+2010-10-02 04:46 +0000 [r289875]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 289874 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
+	  (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010)
+	  | 8 lines When forwarding a message, a prepend means that the
+	  filesystem will always have a better copy. (closes issue #17803)
+	  Reported by: dpetersen Patches: 20100923__issue17803.diff.txt
+	  uploaded by tilghman (license 14) Tested by: dpetersen ........
+	  ................
+
+2010-10-02 02:43 +0000 [r289840]  Jeff Peeler <jpeeler at digium.com>
+
+	* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+	  main/rtp_engine.c, /, channels/chan_sip.c: Merged revisions
+	  289798 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
+	  (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+	  | 15 lines Change RFC2833 DTMF event duration on end to report
+	  actual elapsed time. The scenario here is with a non P2P early
+	  media session. The reported time length of DTMF presses are
+	  coming up short when sending to the remote side. Currently the
+	  event duration is a running total that is incremented when
+	  sending continuation packets. These continuation packets are only
+	  triggered upon incoming media from the remote side, which means
+	  that the running total probably is not going to end up matching
+	  the actual length of time Asterisk received DTMF. This patch
+	  changes the end event duration to be lengthened if it is detected
+	  that the end event is going to come up short. Review:
+	  https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+	  ................
+
+2010-10-01 17:19 +0000 [r289718]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+	  289704 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
+	  (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+	  2010) | 6 lines Disable debugging by default and reformat .config
+	  file. Review: https://reviewboard.asterisk.org/r/929/ ........
+	  ................
+
+2010-10-01 16:22 +0000 [r289701]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 289700 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
+	  (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+	  | 14 lines Ensure user portion of SIP URI matches dialplan when
+	  using encoded characters. This commit takes a simliar approach to
+	  288112 and checks the dialplan to determine the proper action for
+	  an incoming contact header as to whether or not it should be
+	  decoded or not. sip_new was blindly always decoding the
+	  extension, which also caused the outgoing contact header to be
+	  incorrect as well as failing to match the encoded extension in
+	  the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+	  bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+	  wdoekes ........ ................
+
+2010-10-01 09:42 +0000 [r289622]  Stefan Schmidt <sst at sil.at>
+
+	* channels/chan_sip.c: don't iterate through all dialogs to find
+	  and delete old subscribes On every incoming subscribe there is a
+	  iteration through all dialogs to find old subscribes and delete
+	  them. This is slow and not RFC conform. This was only needed in
+	  1.2 cause a subscribe was not deleted when a dialog was
+	  destroyed, after 1.4 a subscribe get removed when its dialog is
+	  destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+	  schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 20:23 +0000 [r289581]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_env.c: Solaris fixes.
+
+2010-09-30 19:53 +0000 [r289554]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 289553 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
+	  2010) | 4 lines Properly handle channel allocation failures duing
+	  invites with replaces. ABE-2588 ........
+
+2010-09-30 19:28 +0000 [r289549]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_misdn.c: Merged revision 289547 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
+	  30 Sep 2010) | 10 lines In chan_misdn, the
+	  DivertingLegInformation2 DivertingNr is garbage when the number
+	  is restricted. The same thing happens with
+	  DivertingLegInformation1 DivertedTo number. The
+	  misdn_PresentedNumberUnscreened_extract() extracted the
+	  Unscreened PartyNumber field unconditionally. It now checks the
+	  presented number unscreened type to see if the PartyNumber was
+	  even present. JIRA ABE-2595 ..........
+
+2010-09-30 17:50 +0000 [r289543]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/localtime.h, main/stdtime/localtime.c,

[... 24768 lines stripped ...]



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