[asterisk-commits] tilghman: tag 1.6.1.5-rc1 r211724 - /tags/1.6.1.5-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Aug 11 16:32:31 CDT 2009


Author: tilghman
Date: Tue Aug 11 16:32:27 2009
New Revision: 211724

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=211724
Log:
Importing files for 1.6.1.5-rc1 release.

Added:
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    tags/1.6.1.5-rc1/.version   (with props)
    tags/1.6.1.5-rc1/ChangeLog   (with props)

Added: tags/1.6.1.5-rc1/.lastclean
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Added: tags/1.6.1.5-rc1/ChangeLog
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--- tags/1.6.1.5-rc1/ChangeLog (added)
+++ tags/1.6.1.5-rc1/ChangeLog Tue Aug 11 16:32:27 2009
@@ -1,0 +1,56863 @@
+2009-08-11  Tilghman Lesher <tlesher at digium.com>
+
+	* Asterisk 1.6.1.5-rc1 released
+
+2009-08-10 19:51 +0000 [r211569-211586]  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
+	  ........ ................
+
+	* channels/chan_iax2.c, main/channel.c, main/cdr.c, res/ael/pval.c,
+	  apps/app_setcallerid.c, main/manager.c, apps/app_rpt.c,
+	  main/asterisk.c, funcs/func_rand.c, apps/app_dahdibarge.c,
+	  res/res_config_pgsql.c, funcs/func_timeout.c,
+	  codecs/codec_speex.c, apps/app_record.c, apps/app_morsecode.c,
+	  main/acl.c, funcs/func_cut.c, cdr/cdr_pgsql.c,
+	  apps/app_followme.c, main/enum.c, res/res_config_sqlite.c,
+	  agi/eagi-sphinx-test.c, main/config.c, channels/misdn_config.c,
+	  channels/chan_dahdi.c, funcs/func_channel.c, apps/app_macro.c,
+	  apps/app_sms.c, pbx/pbx_config.c, apps/app_verbose.c, main/dsp.c,
+	  apps/app_voicemail.c, apps/app_adsiprog.c, funcs/func_speex.c,
+	  channels/chan_sip.c, res/res_limit.c, agi/eagi-test.c,
+	  funcs/func_math.c, channels/chan_agent.c, main/utils.c,
+	  channels/iax2-provision.c, apps/app_talkdetect.c,
+	  main/indications.c, channels/chan_oss.c, main/cli.c,
+	  pbx/pbx_loopback.c, res/res_config_curl.c, channels/chan_misdn.c,
+	  res/res_smdi.c, apps/app_osplookup.c, channels/chan_skinny.c,
+	  pbx/pbx_dundi.c, utils/extconf.c, apps/app_mixmonitor.c,
+	  channels/chan_mgcp.c, main/timing.c, doc/CODING-GUIDELINES,
+	  main/pbx.c, utils/muted.c, apps/app_readfile.c,
+	  apps/app_meetme.c, /, apps/app_privacy.c, apps/app_waituntil.c,
+	  cdr/cdr_adaptive_odbc.c, res/res_http_post.c, pbx/dundi-parser.c,
+	  res/res_musiconhold.c, apps/app_queue.c, main/netsock.c,
+	  utils/frame.c, channels/chan_usbradio.c, funcs/func_enum.c,
+	  channels/chan_phone.c, pbx/pbx_spool.c, apps/app_waitforring.c,
+	  funcs/func_odbc.c, main/features.c, res/res_agi.c,
+	  apps/app_minivm.c, main/http.c, res/snmp/agent.c,
+	  res/res_config_ldap.c, apps/app_chanspy.c, apps/app_stack.c,
+	  res/res_odbc.c, funcs/func_dialplan.c, main/dnsmgr.c,
+	  main/frame.c, apps/app_waitforsilence.c, funcs/func_strings.c,
+	  apps/app_disa.c, apps/app_alarmreceiver.c: AST-2009-005
+
+2009-08-10 14:12 +0000 [r211349]  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 [r211234-211277]  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 20:17 +0000 [r211115]  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 18:19 +0000 [r211047]  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 13:09 +0000 [r210994]  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 21:47 +0000 [r210910-210916]  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 17:48 +0000 [r210819]  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 20:28 +0000 [r210681]  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 [r210567]  Leif Madsen <lmadsen at digium.com>
+
+	* doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
+	  (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+	  | 11 lines Update imapstorage.txt documentation. Updated the
+	  imapstorage.txt documentation to reflect that issues with
+	  c-client versions older than 2007 seem to cause crashing issues
+	  that are not seen with more recent versions. Documentation has
+	  been updated to reflect this. (closes issue #14496) Reported by:
+	  vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+	  uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+	  dbrooks ........ ................
+
+2009-08-04 14:54 +0000 [r210240]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile, /: Merged revisions 210238 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
+	  2009) | 16 lines Merged revisions 210237 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+	  2009) | 10 lines Eliminate spurious compiler warnings from system
+	  headers on *BSD platforms. Ensure that system headers located in
+	  /usr/local/include are actually treated as system headers by the
+	  compiler, and not as local headers which are subject to warnings
+	  from the -Wundef compiler option and others. (closes issue
+	  #15606) Reported by: mvanbaak ........ ................
+
+2009-08-01 11:32 +0000 [r209836-209900]  Russell Bryant <russell at digium.com>
+
+	* main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r209887 | russell | 2009-08-01 06:29:25 -0500
+	  (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+	  | 5 lines Resolve a valgrind warning about a read from
+	  uninitialized memory. (issue #15396) Reported by: aragon ........
+	  ................
+
+	* apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r209839 | russell | 2009-08-01 06:02:07 -0500
+	  (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
+	  | 13 lines Modify how Playtones() is used in Milliwatt() to
+	  resolve gain issue. When Milliwatt() was changed internally to
+	  use Playtones() so that the proper tone was used, it introduced a
+	  drop in gain in the output signal. So, use the playtones API
+	  directly and specify a volume argument such that the output
+	  matches the gain of the original Milliwatt() code. (closes issue
+	  #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
+	  uploaded by russell (license 2) Tested by: rue_mohr ........
+	  ................
+
+	* /, main/event.c: Merged revisions 209835 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
+	  russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
+	  Fix ast_event_queue_and_cache() to actually do the cache() part.
+	  (closes issue #15624) Reported by: ffossard Tested by: russell
+	  ........
+
+2009-08-01 01:25 +0000 [r209781]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile,
+	  channels/misdn/ie.c: Merged revisions 209760-209761 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500
+	  (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+	  2009) | 7 lines Minor changes inspired by testing with latest
+	  GCC. The latest GCC (what will become 4.5.x) has a few new
+	  warnings, that in these cases found some either downright buggy
+	  code, or at least seriously poorly designed code that could be
+	  improved. ........ ................ r209761 | kpfleming |
+	  2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
+	  accidental Makefile change. ................
+
+2009-07-31 21:58 +0000 [r209714]  Russell Bryant <russell at digium.com>
+
+	* /, main/event.c: Merged revisions 209711 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
+	  russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
+	  Fix some places where ast_event_type was used instead of
+	  ast_event_ie_type. ........
+
+2009-07-30 18:46 +0000 [r209593]  David Brooks <dbrooks at digium.com>
+
+	* include/asterisk/abstract_jb.h, channels/chan_dahdi.c,
+	  contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
+	  codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions
+	  209554 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+	  dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+	  Fixes numerous spelling errors. Patch submitted by alecdavis.
+	  (closes issue #15595) Reported by: alecdavis ........
+
+2009-07-30 14:40 +0000 [r209517]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
+	  mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
+	  lines Fix a crash that can result if text codecs are allowed but
+	  textsupport is disabled. (closes issue #15596) Reported by:
+	  fabled Patches: sip-red.patch uploaded by fabled (license 448)
+	  ........
+
+2009-07-28 00:19 +0000 [r209327]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+	  | 9 lines Merged revisions 209315 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+	  | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209262-209281]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+	  kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+	  lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+	  messages about T.38 negotiation in debug level 1 messages, clean
+	  up some looping logic, and correct an improper use of ast_free()
+	  for freeing an ast_frame. ........
+
+	* /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+	  kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+	  lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+	  mode, if the channel that ReceiveFAX is running on supports T.38,
+	  we should *always* attempt to switch T.38, rather than listening
+	  for an incoming CNG tone and only triggering on that. The channel
+	  may be using a low-bitrate codec that distorts the CNG tone, the
+	  sending FAX endpoint may not send CNG at all, or there could be a
+	  variety of other reasons that we don't detect it, but in all
+	  those cases if T.38 is available we certainly want to use it.
+	  ........
+
+2009-07-27 20:57 +0000 [r209237]  Mark Michelson <mmichelson at digium.com>
+
+	* main/rtp.c, /: Merged revisions 209235 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
+	  mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
+	  lines Gracefully handle malformed RTP text packets. AST-2009-004
+	  ........
+
+2009-07-27 20:28 +0000 [r209233]  David Brooks <dbrooks at digium.com>
+
+	* res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
+	  channels/chan_vpb.cc, res/res_smdi.c, /,
+	  include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
+	  revisions 209098 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+	  dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+	  Fixing typos. Replaces "recieved" with "received" and "initilize"
+	  with "initialize" (closes issue #15571) Reported by: alecdavis
+	  ........
+
+2009-07-27 20:17 +0000 [r209134-209199]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+	  2009) | 9 lines Honor channel's music class when using realtime
+	  music on hold. (closes issue #15051) Reported by: alexh Patches:
+	  15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+	  ........
+
+	* main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+	  209132 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+	  2009) | 24 lines Merged revisions 209131 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+	  2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+	  if desired. There are some VoIP providers out there that will not
+	  accept SDP offers with odd numbered UDPTL ports. While it is my
+	  personal opinion that these VoIP providers are misinterpreting
+	  RFC 2327, it really is not a big deal to play along with their
+	  silly little games. Of course, since restricting UDPTL ports to
+	  only even numbers reduces the range of available ports by half,
+	  so the option to use only even port numbers is off by default. A
+	  user can enable the behavior by setting use_even_ports=yes in
+	  udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+	  15182.patch uploaded by mmichelson (license 60) Tested by:
+	  CGMChris ........ ................
+
+2009-07-27 15:40 +0000 [r209058]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile, /: Merged revisions 209056 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+	  kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+	  lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+	  underscore-variants to sub-makes. During the recent Makefile
+	  improvements I made, it seemed the 'make' was automatically
+	  carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+	  I removed the explict export of them. However, there are some
+	  circumstances where make does this, and some where it does not,
+	  so I've brought them back to ensure they are always exported. I
+	  also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+	  platforms. ........
+
+2009-07-27  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.1.2 released
+
+2009-06-05  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.1.1 released
+
+2009-06-04  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Additional updates for AST-2009-001
+
+2009-06-04  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001
+
+2009-04-27  Leif Madsen <lmadsen at digium.com>
+
+	* Create Asterisk 1.6.1.0
+
+2009-04-20  Leif Madsen <lmadsen at digium.com>
+
+	* Create Asterisk 1.6.1.0-rc5
+
+2009-04-20 17:08 +0000 [r189352]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
+	  file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
+	  Fix a bug with non-UDP connections that caused dialogs to not get
+	  freed. This issue crept up because of a reference count issue on
+	  non-UDP based dialogs. The dialog reference count was increased
+	  when transmitting a packet reliably but never decreased. This
+	  caused the dialog structure to hang around despite being unlinked
+	  from the dialogs container. (closes issue #14919) Reported by:
+	  vrban ........
+
+2009-04-20 14:06 +0000 [r189280]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c, /: Merged revisions 189278 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
+	  2009) | 18 lines Merged revisions 189277 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
+	  2009) | 12 lines Move the check for chan->fdno == -1 to after the
+	  zombie/hangup check. Many users were finding that their hung up
+	  channels were staying up and causing 100% CPU usage. (issue
+	  #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+	  uploaded by mmichelson (license 60) Tested by: falves11, bamby
+	  ........ ................
+
+2009-04-18 01:38 +0000 [r189206]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
+	  (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
+	  | 12 lines Fixed autologoff in agents.conf not working when agent
+	  logs in via AgentLogin app An agent logs in by calling an
+	  extension that calls the AgentLogin app. In agents.conf
+	  ackcall=always is set, so when they get a call they have the
+	  choice to either acknowledge it or ignore it. autologoff=10 is
+	  set as well, so if the agent ignores the call over 10sec one may
+	  assume that the agent should be logged out (and in this case
+	  hungup on as well), but this was not happening. (closes issue
+	  #14091) Reported by: evandro Patches: autologoff.diff uploaded by
+	  dvossel (license 671) Review:
+	  http://reviewboard.digium.com/r/225/ ........ ................
+
+2009-04-17 21:55 +0000 [r189139]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+	  revisions 189137 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
+	  | 17 lines Merged revisions 188833,189134 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
+	  | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
+	  Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
+	  rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
+	  Modifed/added some debug messages. JIRA ABE-1835 ........
+	  ................
+
+2009-04-17 20:21 +0000 [r189103]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
+	  mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
+	  lines Prevent a crash when SIP blonde transferring an unbridged
+	  call. If one attempts to use the attended transfer button on a
+	  SIP phone to transfer an unbridged call (such as a call to an
+	  IVR) but hangs up while the target of the transfer is still
+	  ringing, we need to not crash. The problem was that ast_hangup
+	  was called from outside the channel thread. AST-211 ........
+
+2009-04-17 19:46 +0000 [r189080]  Sean Bright <sean.bright at gmail.com>
+
+	* main/asterisk.c, /: Merged revisions 189077 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
+	  seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
+	  line Fix copy/paste error with 'transmit silence' flag. ........
+
+2009-04-17 17:33 +0000 [r189069]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 189010 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
+	  2009) | 12 lines Merged revisions 189009 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
+	  2009) | 5 lines Make Busy() application set the CDR disposition
+	  to BUSY. (closes issue #14306) Reported by: cristiandimache
+	  ........ ................
+
+2009-04-17 14:48 +0000 [r188940-188949]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
+	  22 lines Merged revisions 188946 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
+	  15 lines Fix a bug where a value used to create the channel name
+	  was bogus. This commit fixes the scenario where an incoming call
+	  is authenticated using a peer entry. Previously the channel name
+	  was created using either the username setting from the sip.conf
+	  entry or the IP address that the call came from. Now the channel
+	  name will be created using the peer name itself. This commit will
+	  not change the way the channel name is generated for users or
+	  friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
+	  chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
+	  Nick_Lewis, file ........ ................
+
+	* channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
+	  17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
+	  lines Fix a situation where the DAHDI channel private structure
+	  lock was not unlocked when it should have been. (issue AST-210)
+	  ........ ................
+
+2009-04-16 22:05 +0000 [r188776-188838]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
+	  | 14 lines Merged revisions 188835 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
+	  | 7 lines Only update realtime, if global option rtupdate !=
+	  false (closes issue #14885) Reported by: deepesh Patches:
+	  20090413__bug14885.diff.txt uploaded by tilghman (license 14)
+	  Tested by: deepesh ........ ................
+
+	* apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
+	  (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
+	  | 4 lines Umask should not be exported into global namespace.
+	  (closes issue #14912) Reported by: jcapp ........
+	  ................
+
+2009-04-15 22:12 +0000 [r188649]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
+	  (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
+	  | 12 lines National prefix inserted even when caller ID not
+	  available When the caller ID is restricted, the expected behavior
+	  is for the caller id to be blank. In chan_dahdi, the national
+	  prefix is placed onto the callers number even if its restricted
+	  (empty) causing the caller id to be the national prefix rather
+	  than blank. (closes issue #13207) Reported by: shawkris Patches:
+	  national_prefix.diff uploaded by dvossel (license 671) Review:
+	  http://reviewboard.digium.com/r/220/ ........ ................
+
+2009-04-15 20:20 +0000 [r188473-188596]  Mark Michelson <mmichelson at digium.com>
+
+	* /, main/file.c: Merged revisions 188585 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
+	  2009) | 13 lines Merged revisions 188582 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
+	  2009) | 7 lines Update ast_readvideo_callback to match
+	  ast_readaudio_callback. This fixes potential refcount errors that
+	  may occur on ast_filestreams. AST-208 ........ ................
+
+	* apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
+	  mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
+	  lines Fix a couple of queue member reference leaks. ........
+
+2009-04-14 17:43 +0000 [r188254-188415]  Joshua Colp <jcolp at digium.com>
+
+	* main/rtp.c, /: Merged revisions 188413 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
+	  file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
+	  an incorrect clock rate when sending T140 text. (closes issue
+	  #14029) Reported by: epicac ........
+
+	* /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
+	  file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
+	  a bug with the change I made yesterday to outbound proxy support.
+	  Per discussion with oej on IRC we need the actual IP address, not
+	  the outbound proxy IP address, in the sa field. Upon further
+	  inspection this should make the behaviour of all other uses of
+	  the outbound proxy in the code. ........
+
+2009-04-14 05:46 +0000 [r188208-188212]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c, /: Merged revisions 188210 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
+	  tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
+	  As suggested by Russell, warn users when their dialplan arguments
+	  contain pipes, but not commas. ........
+
+	* /, utils/smsq.c: Merged revisions 188206 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
+	  tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
+	  Application delimiter is ',', not '|'. (closes issue #14881)
+	  Reported by: stegro Patches: smsq.patch uploaded by stegro
+	  (license 752) ........
+
+2009-04-13 19:33 +0000 [r188104]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
+	  2009) | 5 lines Fix another crash related to cached realtime
+	  music on hold. This was another off-by-one problem caused by
+	  moh_register. ........
+
+2009-04-13 16:32 +0000 [r188069]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
+	  file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
+	  Fix a bug where using an outbound proxy would cause the local
+	  address to be 127.0.0.1. Copy the outbound proxy IP address into
+	  the SIP dialog structure as the IP address we will be sending to.
+	  This has to be done because the logic that determines what local
+	  IP address to use in the SIP messages is not aware of an outbound
+	  proxy being in place. It only knows what IP address we are
+	  sending to. (closes issue #12006) Reported by: mnicholson
+	  ........
+
+2009-04-13 14:20 +0000 [r188038]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
+	  mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
+	  lines Set all queue variables on both the caller and member
+	  channels. This allows for the variables to be accessed if a
+	  member macro is run. Thanks to Grigoriy Puzankin for bringing
+	  this up on the -dev list. ........
+
+2009-04-10 20:28 +0000 [r187914]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/Makefile, /: Merged revisions 187906 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
+	  jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
+	  Fix module embedding for chan_h323. Include libchanh323.a in the
+	  modules.link file so that all the symbols can be resolved at link
+	  time. (closes issue #11966) Reported by: dome Patches:
+	  issue_11966.patch uploaded by kpfleming (license 421) Tested by:
+	  jpeeler ........
+
+2009-04-10 17:30 +0000 [r187767]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/scripts/sip-friends.sql,
+	  contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
+	  (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
+	  Apr 2009) | 2 lines Add lastms column to the contributed table
+	  designs ........ ................
+
+2009-04-10 16:54 +0000 [r187723]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, build_tools/embed_modules.xml: Merged revisions 187721 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
+	  Apr 2009) | 5 lines clean up some patterns for files to remove
+	  add embedding support for bridge and test modules ........
+
+2009-04-10 16:03 +0000 [r187678]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
+	  tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
+	  Ensure pvt is not NULL before dereferencing it. (closes issue
+	  #14784) Reported by: pj ........
+
+2009-04-10 16:00 +0000 [r187676]  Russell Bryant <russell at digium.com>
+
+	* tests/test_heap.c, /: Merged revisions 187675 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 |
+	  russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines
+	  Disable test modules by default. ........
+

[... 56164 lines stripped ...]



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