[asterisk-commits] lmadsen: tag 1.6.0.16-rc1 r219207 - /tags/1.6.0.16-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Sep 17 11:02:43 CDT 2009


Author: lmadsen
Date: Thu Sep 17 11:02:38 2009
New Revision: 219207

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=219207
Log:
Importing files for 1.6.0.16-rc1 release.

Added:
    tags/1.6.0.16-rc1/.lastclean   (with props)
    tags/1.6.0.16-rc1/.version   (with props)
    tags/1.6.0.16-rc1/ChangeLog   (with props)

Added: tags/1.6.0.16-rc1/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.16-rc1/.lastclean?view=auto&rev=219207
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Added: tags/1.6.0.16-rc1/.version
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Added: tags/1.6.0.16-rc1/ChangeLog
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.16-rc1/ChangeLog?view=auto&rev=219207
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--- tags/1.6.0.16-rc1/ChangeLog (added)
+++ tags/1.6.0.16-rc1/ChangeLog Thu Sep 17 11:02:38 2009
@@ -1,0 +1,53356 @@
+2009-09-17  Leif Madsen <lmadsen at digium.com>
+
+	* Released Asterisk 1.6.0.16-rc1
+
+2009-09-17 15:44 +0000 [r219198]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/channel.c, /, include/asterisk/cdr.h,
+	  include/asterisk/channel.h: Merged revisions 219139 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
+	  (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+	  2009) | 10 lines Prevent a potential race condition and crash
+	  when hanging up a channel by removing the channel from the
+	  channel list before begining channel tear down. This fix may
+	  potentially cause problems with CDR backends that access the
+	  channel a CDR is associated with via the channel list. This fix
+	  makes the channel unavabile at the time when the CDR backend is
+	  invoked. This has been documented in include/asterisk/cdr.h.
+	  (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+	  Review: https://reviewboard.asterisk.org/r/362/ ........
+	  ................
+
+2009-09-16 23:52 +0000 [r219064]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/config.c, configs/extensions.conf.sample, /: Merged
+	  revisions 219061 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
+	  | 15 lines Merged revisions 219023 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+	  | 8 lines Properly deal with quotes in the arguments of '#exec'
+	  includes. (closes issue #15583) Reported by: pkempgen Patches:
+	  20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+	  20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+	  169) Tested by: pkempgen ........ ................
+
+2009-09-16 19:26 +0000 [r218935]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
+	  mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
+	  lines Reverse order of args to fread. This way, we don't always
+	  write a null byte into byte 1 of the buffer (closes issue #15905)
+	  Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
+	  (license 878) Tested by: ebroad ........
+
+2009-09-16 18:44 +0000 [r218931]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
+	  file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
+	  TCP and TLS connections do not attempt to stop retransmission of
+	  the packet internally. This was preventing responses from being
+	  properly processed because the packet was not being found causing
+	  handle_response to return prematurely. ........
+
+2009-09-16 18:11 +0000 [r218869]  David Brooks <dbrooks at digium.com>
+
+	* main/pbx.c, /: Merged revisions 218868 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
+	  | 20 lines Merged revisions 218867 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+	  | 13 lines Fixes CID pattern matching behavior to mirror that of
+	  extension pattern matching. Pattern matching for extensions uses
+	  a type of scoring system, giving values for specificity to each
+	  character in the pattern. Unfortunately, this is done character
+	  by character, in order. This does lead to some less specific
+	  patterns being first in line for matching, but it will usually
+	  get the job done. This patch merely brings CID matching to the
+	  same level as extension matching. This patch does not attempt to
+	  tackle the problem shared by extension matching. (closes issue
+	  #14708) Reported by: klaus3000 ........ ................
+
+2009-09-16 13:36 +0000 [r218800]  Russell Bryant <russell at digium.com>
+
+	* contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+	  revisions 218799 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
+	  | 16 lines Merged revisions 218798 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+	  | 9 lines Remove the IAXy firmware from Asterisk. The firmware
+	  can now be found on downloads.digium.com, where the rest of our
+	  binary downloads live. This was the last part of our Asterisk
+	  tarballs that was considered non-free by Debian. :-) (closes
+	  issue #15838) Reported by: paravoid ........ ................
+
+2009-09-15 22:39 +0000 [r218732]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
+	  (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
+	  | 6 lines If the user enters the same password as before, don't
+	  signal an error when the change does nothing. (closes issue
+	  #15492) Reported by: cbbs70a Patches:
+	  20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+	  ........ ................
+
+2009-09-15 19:31 +0000 [r218690]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
+	  dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
+	  upward bound checking for port string to int conversion ........
+
+2009-09-15 16:21 +0000 [r218601]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
+	  2009) | 15 lines Merged revisions 218578 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
+	  2009) | 8 lines Send request contact header field with response
+	  to registrer queries instead of the address of record. (closes
+	  issue #14438) Reported by: ravindrad Patches: regquerypatch
+	  uploaded by ravindrad (license 684) Tested by: ravindrad ........
+	  ................
+
+2009-09-15 16:05 +0000 [r218580]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_followme.c: Merged revisions 218579 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
+	  | 16 lines Merged revisions 218577 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
+	  | 9 lines Ensure FollowMe sets language in channels it creates.
+	  Also, not in the original bug report, but related fields are
+	  accountcode and musicclass, and the inheritance of datastores.
+	  (closes issue #15372) Reported by: Romik Patches:
+	  20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+	  Tested by: cervajs ........ ................
+
+2009-09-15 15:42 +0000 [r218505-218573]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
+	  mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
+	  lines Use a better method of ensuring null-termination of the
+	  buffer while reading the SDP when using TCP. ........
+
+	* /, channels/chan_sip.c: Merged revisions 218499,218504 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
+	  15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
+	  over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
+	  -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
+	  socket is null-terminated. ........
+
+2009-09-15 15:03 +0000 [r218501]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, sounds/Makefile: Merged revisions 218500 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
+	  2009) | 9 lines Merged revisions 218497 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
+	  2009) | 1 line Use proper hostname for downloading sound files.
+	  ........ ................
+
+2009-09-14 22:49 +0000 [r218431]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Merged revisions 218430 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009)
+	  | 18 lines Merged revisions 218401 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
+	  | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
+	  crash in do_monitor. After talking to rmudgett about some of his
+	  recent iflist locking changes, it was determined that the only
+	  place that would destroy a channel without being explicitly to do
+	  so was in handle_init_event. The loop to walk the interface list
+	  has been modified to wait to destroy the channel until the
+	  dahdi_pvt of the channel to be destroyed is no longer needed.
+	  (closes issue #15378) Reported by: samy ........ ................
+
+2009-09-14 19:49 +0000 [r218362]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /, configs/voicemail.conf.sample,
+	  sounds/Makefile: Merged revisions 218361 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009)
+	  | 11 lines Recorded merge of revisions 218331 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
+	  | 4 lines Don't say "Please try again" if we don't give the user
+	  another chance to try again. (issue #15055, SWP-129) Reported by:
+	  jthurman ........ ................
+
+2009-09-14 15:22 +0000 [r218244]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, apps/app_directed_pickup.c: Merged revisions 218224 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
+	  (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
+	  2009) | 8 lines Ensure we don't pickup ourselves when doing
+	  pickup by exten. (closes issue #15100) Reported by: lmsteffan
+	  Patches: (modified) pickup.patch uploaded by lmsteffan (license
+	  779) ........ ................
+
+2009-09-13 19:10 +0000 [r218216]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
+	  that annoys gcc This memset doesn't write beyond the end of the
+	  buffer. (tmpbuf has size of 4). Merged revisions 218184 via
+	  svnmerge from http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-12 13:10 +0000 [r218108]  Michiel van Baak <michiel at vanbaak.info>
+
+	* main/rtp.c: Use the ip for the new 'rtp set debug ip <foo>'.
+	  Since 1.6.X still has the deprecated 'rtp debug ip <foo>' this
+	  patch is different from the fix that went into trunk (closes
+	  issue #15711) Reported by: davidw Patches:
+	  2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
+	  Tested by: davidw
+
+2009-09-11 05:58 +0000 [r217920-218051]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_queue.c: Merged revisions 217990 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
+	  | 10 lines Merged revisions 217989 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
+	  | 3 lines Don't ring another channel, if there's not enough time
+	  for a queue member to answer. (Fixes AST-228) ........
+	  ................
+
+	* contrib/scripts/iax-friends.sql, /, channels/chan_sip.c,
+	  channels/chan_iax2.c: Merged revisions 217916 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
+	  tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
+	  Make calltoken support work with realtime users and peers.
+	  ........
+
+2009-09-10 22:31 +0000 [r217858-217913]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: sip peer matching by address only with
+	  TCP/TLS This patch removes the contact header matching logic and
+	  adds logic to match all tcp/tls connections by ip only. Thanks to
+	  oej for finding the issue and suggesting solutions. Review:
+	  https://reviewboard.asterisk.org/r/355/
+
+	* /, channels/chan_iax2.c: Merged revisions 217807 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
+	  (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
+	  | 22 lines IAX2 encryption regression The IAX2 Call Token
+	  security patch inadvertently broke the use of encryption due to
+	  the reorganization of code in the socket_process() function. When
+	  encryption is used, an incoming full frame must first be
+	  decrypted before the information elements can be parsed. The
+	  security release mistakenly moved IE parsing before decryption in
+	  order to process the new Call Token IE. To resolve this,
+	  decryption of full frames is once again done before looking into
+	  the frame. This involves searching for an existing callno,
+	  checking the pvt to see if encryption is turned on, and
+	  decrypting the packet before the internal fields of the full
+	  frame are accessed. (closes issue #15834) Reported by: karesmakro
+	  Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
+	  (license 671) Tested by: dvossel, karesmakro Review:
+	  https://reviewboard.asterisk.org/r/355/ ........ ................
+
+2009-09-10 19:53 +0000 [r217736]  mnick <mnick at localhost>:
+
+	* /, res/res_musiconhold.c: Merged revisions 217730 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
+	  17 lines Sets the correct musicclass after an announcement
+	  (closes issue #15279) Reported by: mbeckwell Patches: patch.txt
+	  uploaded by mnick (license ) Tested by: mnick (closes issue
+	  #15832) Reported by: mbeckwell Patches: patch.txt uploaded by
+	  mnick (license 874) Tested by: mnick ........
+
+2009-09-10 12:16 +0000 [r217596]  Olle Johansson <oej at edvina.net>
+
+	* /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
+	  oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
+	  Include ActionID in all events that are responsed to AMI Action
+	  SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
+	  Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
+	  bellamy (license 299) ........
+
+2009-09-09 20:15 +0000 [r217484]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
+	  4.4 has more strict rules for aliasing. It doesn't like a struct
+	  sockaddr_in pointer pointing to a struct sockaddr. So we make it
+	  a union. Merged revisions 217445 via svnmerge from
+	  http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-09 11:33 +0000 [r217405]  Olle Johansson <oej at edvina.net>
+
+	* /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
+	  oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
+	  having any TLS session to write to is a serious XMIT_ERROR.
+	  ........
+
+2009-09-08 21:45 +0000 [r217281]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configure: Commit regenerated configure script that I missed
+	  earlier.
+
+2009-09-08 20:31 +0000 [r217209]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
+	  | 14 lines Merged revisions 217156 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
+	  | 7 lines When MOH is playing on the channel, announcements sent
+	  through the conference are not heard. (closes issue #14588)
+	  Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
+	  uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
+	  tilghman ........ ................
+
+2009-09-08 16:38 +0000 [r217075]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, include/asterisk/autoconfig.h.in, configure.ac: Merged
+	  revisions 217074 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
+	  kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
+	  lines Ensure that the default autoconf CFLAGS are not used. A
+	  recent change to the configure script that allows the user to
+	  specify CFLAGS and/or LDFLAGS to the script had the unfortunate
+	  side effect of letting autoconf's default CFLAGS (-g -O2) feed in
+	  to the rest of the build system, thereby overriding the
+	  DONT_OPTIMIZE setting in menuselect. That problem is now
+	  corrected. ........
+
+2009-09-08 15:35 +0000 [r217034]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, res/res_limit.c: Merged revisions 217033 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
+	  tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
+	  Remove what appears to be an unnecessary define. (closes issue
+	  #15851) Reported by: tzafrir ........
+
+2009-09-08 14:28 +0000 [r216996]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
+	  dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
+	  caller id number empty parse_uri was not being given the correct
+	  scheme's, as a result, uri parsing did not parse the username
+	  correctly. One of the side effects of this is an empty caller id.
+	  (closes issue #15839) Reported by: ebroad Patches:
+	  blank_cidv2.patch uploaded by ebroad (license 878)
+	  parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
+	  ebroad, dvossel ........
+
+2009-09-07 16:38 +0000 [r216645-216843]  Olle Johansson <oej at edvina.net>
+
+	* /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
+	  oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+	  Make sure we reset global_exclude_static at channel reload
+	  ........
+
+	* /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
+	  oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
+	  there is no session timer in the INVITE, set it to default value
+	  (not unset minimum = -1) Patch by oej closes issue #15621
+	  Reported by: fnordian Tested by: atis ........
+
+	* configs/sip.conf.sample: fix documentation so it agrees with code
+
+	* channels/chan_sip.c, CHANGES: Add doc and turn off premature
+	  media filter by default
+
+	* apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c,
+	  apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
+	  04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
+	  lines Make apps send PROGRESS control frame for early media and
+	  fix too early media issue in SIP The issue at hand is that some
+	  legacy (dying) PBX systems send empty media frames on PRI links
+	  *before* any call progress. The SIP channel receives these frames
+	  and by default signals 183 Session progress and starts sending
+	  media. This will cause phones to play silence and ignore the
+	  later 180 ringing message. A bad user experience. The fix is
+	  twofold: - We discovered that asterisk apps that support early
+	  media ("noanswer") did not send any PROGRESS frame to indicate
+	  early media. Fixed. - We introduce a setting in chan_sip so that
+	  users can disable any relay of media frames before the outbound
+	  channel actually indicates any sort of call progress. In 1.4,
+	  1.6.0 and 1.6.1, this will be disabled for backward
+	  compatibility. In later versions of Asterisk, this will be
+	  enabled. We don't assume that it will change your Asterisk phone
+	  experience - only for the better. We encourage third-party
+	  application developers to make sure that if they have
+	  applications that wants to send early media, add a PROGRESS
+	  control frame transmission to make sure that all channel drivers
+	  actually will start sending early media. This has not been the
+	  default in Asterisk previous to this patch, so if you got
+	  inspiration from our code, you need to update accordingly. Sorry
+	  for the trouble and thanks for your support. This code has been
+	  running for a few months in a large scale installation (over 250
+	  servers with PRI and/or BRI links to old PBX systems). That's no
+	  proof that this is an excellent patch, but, well, it's tested :-)
+	  ........ ................
+
+2009-09-04 19:32 +0000 [r216595]  Sean Bright <sean at malleable.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
+	  2009) | 1 line Use ast_free() instead of free(). ........
+
+2009-09-04 17:32 +0000 [r216548]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
+	  Sep 2009) | 3 lines Enable turning off the application delimiter
+	  warning with the 'dontwarn' option. Suggested on the -dev list,
+	  and implemented in an alternate way by me. ........
+
+2009-09-04 15:07 +0000 [r216507]  Michiel van Baak <michiel at vanbaak.info>
+
+	* /, main/utils.c: Merged revisions 216506 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
+	  | 9 lines Merged revisions 216435 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
+	  | 2 lines make asterisk compile under devmode with DEBUG_THREADS
+	  enabled on OpenBSD ........ ................
+
+2009-09-04 10:49 +0000 [r216265]  Russell Bryant <russell at digium.com>
+
+	* doc/IAX2-security.txt (added), /: Merged revisions 216264 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r216264 | russell | 2009-09-04 05:48:44 -0500
+	  (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ................ r216263 | russell | 2009-09-04 05:48:00 -0500
+	  (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
+	  ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
+	  Sep 2009) | 2 lines Add a plain text version of the IAX2 security
+	  document. ........ ................ ................
+
+2009-09-04 06:11 +0000 [r216223]  Michiel van Baak <michiel at vanbaak.info>
+
+	* main/astobj2.c, /: Merged revisions 216222 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
+	  mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
+	  make sure 'start' is always initialized. Makes asterisk compile
+	  with --enable-dev-mode ........
+
+2009-09-03 19:42 +0000 [r216011-216097]  Russell Bryant <russell at digium.com>
+
+	* UPGRADE.txt: tweak
+
+	* /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
+	  | 16 lines Merged revisions 216085 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ................ r216085 | russell | 2009-09-03 14:36:46 -0500
+	  (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
+	  ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
+	  Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
+	  ........ ................ ................
+
+	* /, doc/IAX2-security.pdf (added): Merged revisions 216009 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r216009 | russell | 2009-09-03 13:45:54 -0500
+	  (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ................ r216008 | russell | 2009-09-03 13:44:58 -0500
+	  (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
+	  ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
+	  Sep 2009) | 2 lines Add IAX2 security document related to
+	  AST-2009-006. ........ ................ ................
+
+2009-09-03 18:40 +0000 [r216003]  David Vossel <dvossel at digium.com>
+
+	* channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample,
+	  include/asterisk/acl.h, channels/iax2-parser.h, /,
+	  include/asterisk/astobj2.h, channels/iax2.h, main/acl.c,
+	  channels/chan_iax2.c: Merged revisions 215955 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 |
+	  dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines
+	  Merge code associated with AST-2009-006 (closes issue #12912)
+	  Reported by: rathaus Tested by: tilghman, russell, dvossel,
+	  dbrooks ........
+
+2009-09-03  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.0.15 released
+
+	* AST-2009-006
+
+2009-08-28  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.0.14 released
+
+2009-08-11  Tilghman Lesher <tlesher at digium.com>
+
+	* Released 1.6.0.14-rc1
+
+2009-08-10 19:51 +0000 [r211551-211587]  Tilghman Lesher <tlesher at digium.com>
+
+	* doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+	  (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+	  Aug 2009) | 1 line Conversion specifiers, not format specifiers
+	  ........ ................
+
+	* res/res_config_curl.c, apps/app_waitforring.c,
+	  channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c,
+	  pbx/pbx_config.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+	  res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c,
+	  doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
+	  main/utils.c, cdr/cdr_pgsql.c, res/res_musiconhold.c,
+	  apps/app_followme.c, channels/misdn_config.c, utils/frame.c,
+	  main/channel.c, main/cdr.c, res/ael/pval.c, funcs/func_enum.c,
+	  channels/chan_phone.c, apps/app_osplookup.c,
+	  apps/app_setcallerid.c, main/manager.c, funcs/func_odbc.c,
+	  apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c,
+	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+	  funcs/func_dialplan.c, main/dnsmgr.c, channels/chan_sip.c,
+	  res/res_limit.c, apps/app_waitforsilence.c, agi/eagi-test.c,
+	  apps/app_waituntil.c, main/acl.c, apps/app_queue.c,
+	  channels/chan_oss.c, agi/eagi-sphinx-test.c,
+	  channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c,
+	  apps/app_sms.c, utils/extconf.c, apps/app_verbose.c,
+	  apps/app_stack.c, apps/app_dahdibarge.c, funcs/func_rand.c,
+	  apps/app_readfile.c, main/frame.c, /, apps/app_record.c,
+	  funcs/func_strings.c, cdr/cdr_adaptive_odbc.c,
+	  apps/app_alarmreceiver.c, channels/chan_iax2.c,
+	  main/indications.c, main/config.c, main/cli.c,
+	  pbx/pbx_loopback.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+	  res/res_smdi.c, channels/chan_skinny.c, main/features.c,
+	  main/http.c, main/pbx.c, apps/app_privacy.c,
+	  codecs/codec_speex.c, funcs/func_math.c, channels/chan_agent.c,
+	  apps/app_morsecode.c, apps/app_disa.c, funcs/func_cut.c,
+	  channels/iax2-provision.c, pbx/dundi-parser.c,
+	  apps/app_talkdetect.c: AST-2009-005
+
+2009-08-10 14:10 +0000 [r211348]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+	  file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+	  retrieval of the port used for the video stream when adding SDP
+	  to a SIP message. (closes issue #15121) Reported by: jsmith
+	  ........
+
+2009-08-09 15:43 +0000 [r211233-211276]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/astfd.c: Merged revisions 211275 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+	  | 9 lines Merged revisions 211274 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+	  | 2 lines Small oops. Clear the flags which have been checked.
+	  ........ ................
+
+	* apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+	  tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+	  Check for NULL frame, before dereferencing pointer. (closes issue
+	  #15617) Reported by: rain ........
+
+2009-08-07 20:14 +0000 [r211114]  Russell Bryant <russell at digium.com>
+
+	* apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+	  | 11 lines Recorded merge of revisions 211112 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+	  | 4 lines Resolve a deadlock involving app_chanspy and
+	  masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:18 +0000 [r211044]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_queue.c: Merged revisions 211040 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+	  | 21 lines Merged revisions 211038 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+	  | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+	  not the membername. This is a partial revert of revision 82590,
+	  which was an attempted cleanup, but in reality, it broke
+	  QUEUE_MEMBER_LIST, which has always been intended as a method by
+	  which component interfaces could be queried from the queue.
+	  Membername isn't useful here, because that field cannot be used
+	  to obtain further information about the member. See the
+	  documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+	  QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+	  member argument for further justification. (closes issue #15664)
+	  Reported by: rain Patches: app_queue-queue_member_list.diff
+	  uploaded by rain (license 327) ........ ................
+
+2009-08-07 13:08 +0000 [r210993]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/udptl.c, /: Merged revisions 210992 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+	  kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+	  lines Workaround broken T.38 endpoints that offer tiny
+	  MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+	  the maximum IFP size that should be sent to them, rather than the
+	  maximum packet payload size. If such an endpoint also requests
+	  UDPRedundancy as the error correction mode, we'll end up
+	  calculating a tiny maximum IFP size, so small as to be unusable.
+	  This patch sets a lower bound on what we'll consider the remote's
+	  maximum IFP size to be, assuming that endpoints that do this
+	  really can accept larger packets than they've offered to accept.
+	  (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:46 +0000 [r210909-210915]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c, /: Merged revisions 210914 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+	  | 14 lines Merged revisions 210913 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+	  | 7 lines Because channel information can be accessed outside of
+	  the channel thread, we must lock the channel prior to modifying
+	  it. (closes issue #15397) Reported by: caspy Patches:
+	  20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+	  Tested by: caspy ........ ................
+
+	* apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+	  revisions 210908 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+	  tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+	  Allow Gosub to recognize quote delimiters without consuming them.
+	  (closes issue #15557) Reported by: rain Patches:
+	  20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+	  Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+	  ........
+
+2009-08-06 17:47 +0000 [r210818]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+	  file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+	  Accept additional T.38 reinvites after an initial one has been
+	  handled. Discussion of this subject has yielded that it is not
+	  actually acceptable to change T.38 parameters after the initial
+	  reinvite but declining is harsh and can cause the fax to fail
+	  when it may be possible to allow it to continue. This patch
+	  changes things so that additional T.38 reinvites are accepted but
+	  parameter changes ignored. This gives the fax a fighting chance.
+	  (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:07 +0000 [r210647]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+	  (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+	  | 14 lines Dialplan starts execution before the channel setup is
+	  complete. * Issue 15655: For the case where dialing is complete
+	  for an incoming call, dahdi_new() was asked to start the PBX and
+	  then the code set more channel variables. If the dialplan hungup
+	  before these channel variables got set, asterisk would likely
+	  crash. * Fixed potential for overlap incoming call to erroneously
+	  set channel variables as global dialplan variables if the
+	  ast_channel structure failed to get allocated. * Added missing
+	  set of CALLINGSUBADDR in the dialing is complete case. (closes
+	  issue #15655) Reported by: alecdavis ........ ................
+
+2009-08-05 18:57 +0000 [r210568]  Leif Madsen <lmadsen at digium.com>
+
+	* doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge

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