[asterisk-commits] lmadsen: tag 1.6.0.11-rc1 r202930 - /tags/1.6.0.11-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jun 24 13:16:11 CDT 2009


Author: lmadsen
Date: Wed Jun 24 13:16:07 2009
New Revision: 202930

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=202930
Log:
Importing files for 1.6.0.11-rc1 release.

Added:
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    tags/1.6.0.11-rc1/.version   (with props)
    tags/1.6.0.11-rc1/ChangeLog   (with props)

Added: tags/1.6.0.11-rc1/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.11-rc1/.lastclean?view=auto&rev=202930
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Added: tags/1.6.0.11-rc1/ChangeLog
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--- tags/1.6.0.11-rc1/ChangeLog (added)
+++ tags/1.6.0.11-rc1/ChangeLog Wed Jun 24 13:16:07 2009
@@ -1,0 +1,53308 @@
+2009-06-24  Leif Madsen <lmadsen at digium.com>
+
+	* Released Asterisk 1.6.0.11-rc1
+
+2009-06-24 18:09 +0000 [r202926]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 |
+	  file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
+	  Ensure the default settings are applied for T.38 when we set it
+	  up for a peer. ........
+
+2009-06-23 22:09 +0000 [r202763]  Matthew Fredrickson <creslin at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) |
+	  1 line I could have sworn I committed this patch ages ago, but...
+	  bug fix with setting NAI properly on linksets in certain
+	  situations. ........
+
+2009-06-23 21:26 +0000 [r202754]  Ryan Brindley <rbrindley at digium.com>
+
+	* main/config.c, /: Merged revisions 202753 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 |
+	  rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9
+	  lines If we delete the info, lets also delete the lines (closes
+	  issue #14509) Reported by: timeshell Patches:
+	  20090504__bug14509.diff.txt uploaded by tilghman (license 14)
+	  Tested by: awk, timeshell ........
+
+2009-06-23 16:40 +0000 [r202675]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009)
+	  | 18 lines Merged revisions 202671 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
+	  | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
+	  non-standard port and transport (closes issue #14659) Reported
+	  by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
+	  by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
+	  by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
+	  https://reviewboard.asterisk.org/r/288/ ........ ................
+
+2009-06-22 20:12 +0000 [r202498]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 202497 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009)
+	  | 11 lines Merged revisions 202496 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
+	  | 4 lines Report CallerID change during a masquerade. Reported
+	  by: markster ........ ................
+
+2009-06-22 16:30 +0000 [r202471]  Sean Bright <sean at malleable.com>
+
+	* cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun
+	  2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid
+	  potential crashes during reload. Pointed out by Russell while
+	  working on the CEL branch. ........
+
+2009-06-22 16:06 +0000 [r202416]  Russell Bryant <russell at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009)
+	  | 9 lines Merged revisions 202414 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
+	  | 2 lines Make Polycom subscription type override check more
+	  explicit. ........ ................
+
+2009-06-22 15:05 +0000 [r202338-202344]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun
+	  2009) | 36 lines Merged revisions 202341-202342 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
+	  2009) | 26 lines Fix a situation in which Asterisk would not stop
+	  retransmitting 487s. If a CANCEL were received by Asterisk, we
+	  would send a 487 in response to the original INVITE and a 200 OK
+	  for the CANCEL. If there were a network hiccup which caused the
+	  200 OK and the 487 to be lost, then the UA communicating with
+	  Asterisk may try to retransmit its CANCEL. Asterisk's response to
+	  this used to be to try sending another 487 to the canceled INVITE
+	  and another 200 OK to the CANCEL. The problem here is that the
+	  originally-sent 487 was sent "reliably" meaning that it will be
+	  retransmitted until it is received properly. So when we receive
+	  the second CANCEL it is likely that the first batch of 487s we
+	  sent is still going strong and reaches the UA. The result was
+	  that the second set of 487s would be retransmitted constantly
+	  until the maximum number of retries had been reached. The fix for
+	  this is that if we receive a second CANCEL for an INVITE, then we
+	  cancel the retransmission of the first set of 487s and start a
+	  second set. This causes the dialog to be terminated reasonably.
+	  (closes issue #14584) Reported by: klaus3000 Patches:
+	  14584_v2.patch uploaded by mmichelson (license 60) Tested by:
+	  klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
+	  -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
+	  left from previous commit. ........ ................
+
+	* /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun
+	  2009) | 31 lines Merged revisions 202336 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
+	  2009) | 25 lines Fix a possible infinite loop in SDP parsing
+	  during glare situation. There was a while loop in
+	  get_ip_and_port_from_sdp which was controlled by a call to
+	  get_sdp_iterate. The loop would exit either if what we were
+	  searching for was found or if the return was NULL. The problem is
+	  that get_sdp_iterate never returns NULL. This means that if what
+	  we were searching for was not present, the loop would run
+	  infinitely. This modification of the loop fixes the problem.
+	  (closes issue #15213) Reported by: schmidts (closes issue #15349)
+	  Reported by: samy (closes issue #14464) Reported by: pj (closes
+	  issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
+	  uploaded by mmichelson (license 60) Tested by: aragon ........
+	  ................
+
+2009-06-21 16:14 +0000 [r202259-202263]  Russell Bryant <russell at digium.com>
+
+	* cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 |
+	  russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
+	  Fix possibility of crashiness during reload in custom fields
+	  handling. ........
+
+	* cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 |
+	  russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
+	  Standardize return values of load_config() so reload() doesn't
+	  report an error on success. ........
+
+2009-06-20 19:14 +0000 [r202184]  Sean Bright <sean at malleable.com>
+
+	* /, apps/app_fax.c: Merged revisions 202183 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 |
+	  seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5
+	  lines Fix version detection for API changes in spandsp. (closes
+	  issue #15355) Reported by: deuffy ........
+
+2009-06-19 21:07 +0000 [r202006]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Added deadlock protection to
+	  try_suggested_sip_codec in chan_sip.c. Review:
+	  https://reviewboard.asterisk.org/r/287/
+
+2009-06-19 20:27 +0000 [r201997]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_iax2.c: Merged revisions 201994 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500
+	  (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009)
+	  | 8 lines timestamp was being converted to host order as a short
+	  rather than a long (closes issue #15361) Reported by: ffloimair
+	  Patches: ts_issue.diff uploaded by dvossel (license 671) ........
+	  ................
+
+2009-06-19 00:44 +0000 [r201786-201830]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/features.c: Merged revisions 201829 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009)
+	  | 13 lines Merged revisions 201828 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
+	  | 6 lines If the "h" extension fails, give it another chance in
+	  main/pbx.c. If the "h" extension fails, give it another chance in
+	  main/pbx.c, when it returns from the bridge code. Fixes an issue
+	  where the "h" extension may occasionally not fire, when a Dial is
+	  executed from a Macro. Debugged in #asterisk with user tompaw.
+	  ........ ................
+
+	* /, apps/Makefile: Merged revisions 201783 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 |
+	  tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
+	  One of the changes in 1.6.1 was to allow app_directory to use
+	  functionality within app_voicemail for directory functions. It is
+	  therefore no longer necessary for app_directory to be linked
+	  against the ODBC libraries (and it never was necessary for
+	  app_directory to be linked against IMAP, though it was). ........
+
+2009-06-18 16:58 +0000 [r201682]  David Vossel <dvossel at digium.com>
+
+	* channels/misdn/isdn_lib.c, main/asterisk.c, utils/conf2ael.c,
+	  main/ast_expr2.c, utils/stereorize.c,
+	  codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx,
+	  main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c,
+	  utils/extconf.c, pbx/pbx_config.c, res/res_config_ldap.c: Merged
+	  revisions 201678 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 |
+	  dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines
+	  fixes some memory leaks and redundant conditions (closes issue
+	  #15269) Reported by: contactmayankjain Patches: patch.txt
+	  uploaded by contactmayankjain (license 740)
+	  memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
+	  Tested by: contactmayankjain, dvossel ........
+
+2009-06-18 15:32 +0000 [r201612]  Russell Bryant <russell at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r201610 | russell | 2009-06-18 10:27:10 -0500
+	  (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009)
+	  | 29 lines Fix memory corruption and leakage related reloads of
+	  non files mode MoH classes. For Music on Hold classes that are
+	  not files mode, meaning that we are executing an application that
+	  will feed us audio data, we use a thread to monitor the external
+	  application and read audio from it. This thread also makes use of
+	  the MoH class object. In the MoH class destructor, we used
+	  pthread_cancel() to ask the thread to exit. Unfortunately, the
+	  code did not wait to ensure that the thread actually went away.
+	  What needed to be done is a pthread_join() to ensure that the
+	  thread fully cleans up before we proceed. By adding this one
+	  line, we resolve two significant problems: 1) Since the thread
+	  was never joined, it never fully goes away. So, on every reload
+	  of non-files mode MoH, an unused thread was sticking around. 2)
+	  There was a race condition here where the application monitoring
+	  thread could still try to access the MoH class, even though the
+	  thread executing the MoH reload has already destroyed it. (issue
+	  #15109) Reported by: jvandal (issue #15123) Reported by:
+	  axisinternet (issue #15195) Reported by: amorsen (issue AST-208)
+	  ........ ................
+
+2009-06-17 20:10 +0000 [r201459-201463]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 |
+	  mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12
+	  lines Fix problem with no audio due to ignoring the SDP. A recent
+	  change to our SDP version comparison made audio not function on
+	  some calls. This was because of a test wherein we were trying to
+	  see if an unsigned value was less than 0. This is a dumb
+	  comparison and arguably the compiler should have warned about it.
+	  Alas, though, it slipped past. Now it's fixed by changing the
+	  variable to be a signed type. Found by several developers. Tested
+	  by mnicholson and dbrooks. ........
+
+	* main/channel.c, /: Merged revisions 201458 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun
+	  2009) | 15 lines Merged revisions 201450 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
+	  2009) | 9 lines Change the datastore traversal in
+	  ast_do_masquerade to use a safe list traversal. It is possible
+	  for datastore fixup functions to remove the datastore from the
+	  list and free it. In particular, the queue_transfer_fixup in
+	  app_queue does this. While I don't yet know of this causing any
+	  crashes, it certainly could. Found while discussing a separate
+	  issue with Brian Degenhardt. ........ ................
+
+2009-06-17 19:55 +0000 [r201449]  David Vossel <dvossel at digium.com>
+
+	* apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500
+	  (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009)
+	  | 19 lines StopMixMonitor race condition (not giving up file
+	  immediately) StopMixMonitor only indicates to the MixMonitor
+	  thread to stop writing to the file. It does not guarantee that
+	  the recording's file handle is available to the dialplan
+	  immediately after execution. This results in a race condition. To
+	  resolve this, the filestream pointer is placed in a datastore on
+	  the channel. When StopMixMonitor is called, the datastore is
+	  retrieved from the channel and the filestream is closed
+	  immediately before returning to the dialplan. Documentation
+	  indicating the use of StopMixMonitor to free files has been
+	  updated as well. (closes issue #15259) Reported by: travisghansen
+	  Tested by: dvossel Review:
+	  https://reviewboard.asterisk.org/r/283/ ........ ................
+
+2009-06-17 19:35 +0000 [r201443]  David Brooks <dbrooks at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009)
+	  | 16 lines Merged revisions 201380 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
+	  | 9 lines Checks for NULL sip_pvt pointer in
+	  chan_sip.c->acf_channel_read() Zombie channels could be passed,
+	  and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
+	  checking for NULL pointer. (closes issue #15330) Reported by:
+	  okrief Tested by: dbrooks ........ ................
+
+2009-06-17 12:05 +0000 [r201263]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, include/asterisk/linkedlists.h: Merged revisions 201262 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500
+	  (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
+	  2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
+	  to be appended is empty. When the list to be appended is empty,
+	  and the list to be appended to is *not*, AST_LIST_APPEND_LIST
+	  would actually cause the target list to become broken, and no
+	  longer have a pointer to its last entry. This patch fixes the
+	  problem. (reported by Stanislaw Pitucha on the asterisk-dev
+	  mailing list) ........ ................
+
+2009-06-16 22:31 +0000 [r201226]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 |
+	  dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
+	  fix issue with build_contact introduced by the "SIP trasnport
+	  type issues" commit ........
+
+2009-06-16 19:34 +0000 [r201093]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c,
+	  main/autoservice.c, main/frame.c, /, apps/app_meetme.c,
+	  configure, main/slinfactory.c, autoconf/ast_gcc_attribute.m4,
+	  configure.ac, include/asterisk/linkedlists.h, main/file.c,
+	  include/asterisk/channel.h, include/asterisk/frame.h: Merged
+	  revisions 201056,201090 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun
+	  2009) | 18 lines Merged revisions 200991 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
+	  2009) | 11 lines Improve support for media paths that can
+	  generate multiple frames at once. There are various media paths
+	  in Asterisk (codec translators and UDPTL, primarily) that can
+	  generate more than one frame to be generated when the application
+	  calling them expects only a single frame. This patch addresses a
+	  number of those cases, at least the primary ones to solve the
+	  known problems. In addition it removes the broken TRACE_FRAMES
+	  support, fixes a number of bugs in various frame-related API
+	  functions, and cleans up various code paths affected by these
+	  changes. https://reviewboard.asterisk.org/r/175/ ........
+	  ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500
+	  (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler
+	  attribute checking. Defaulting to 'static' for the function scope
+	  was bad... so remove it. ................
+
+2009-06-16 17:11 +0000 [r200992]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 |
+	  dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
+	  SIP transport type issues What this patch addresses: 1.
+	  ast_sip_ouraddrfor() by default binds to the UDP address/port
+	  reguardless if the sip->pvt is of type UDP or not. Now when no
+	  remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
+	  transport type, attempting to set the address and port to the
+	  correct TCP/TLS bindings if necessary. 2. It is not necessary to
+	  send the port number in the Contact header unless the port is
+	  non-standard for the transport type. This patch fixes this and
+	  removes the todo note. 3. In sip_alloc(), the default dialog
+	  built always uses transport type UDP. Now sip_alloc() looks at
+	  the sip_request (if present) and determines what transport type
+	  to use by default. 4. When changing the transport type of a
+	  sip_socket, the file descriptor must be set to -1 and in some
+	  cases the tcptls_session's ref count must be decremented and set
+	  to NULL. I've encountered several issues associated with this
+	  process and have created a function, set_socket_transport(), to
+	  handle the setting of the socket type. (closes issue #13865)
+	  Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
+	  Kristijan (license 753) 13865.patch uploaded by mmichelson
+	  (license 60) tls_port_v5.patch uploaded by vrban (license 756)
+	  transport_issues.diff uploaded by dvossel (license 671) Tested
+	  by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
+	  https://reviewboard.asterisk.org/r/278/ ........
+
+2009-06-16 16:34 +0000 [r200986]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+	  revisions 200985 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 |
+	  kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7
+	  lines Fix problems with new compiler attribute checking in
+	  configure script. The last changes to ast_gcc_attribute.m4 caused
+	  some problems checking for various attributes, because the scope
+	  of the symbol the attribute is applied to can be important; this
+	  patch allows the scope to be specified for the check. ........
+
+2009-06-16 16:02 +0000 [r200945]  Michiel van Baak <michiel at vanbaak.info>
+
+	* apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009)
+	  | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail
+	  can only use one storage module at the moment. Because it's
+	  unclear that selecting one of the storage modules in menuselect
+	  will disable filesystem storage we now have a FILE_STORAGE option
+	  that conflicts with the other modules. (closes issue #15333)
+	  ........
+
+2009-06-16 01:33 +0000 [r200724-200767]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, configure, include/asterisk/autoconfig.h.in,
+	  autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15
+	  Jun 2009) | 11 lines Ensure that configure-script testing for
+	  compiler attributes actually works. The configure script tests
+	  for compiler attributes didn't actually enable enough warnings or
+	  provide a proper test harness to determine whether the compiler
+	  supports the attribute in question or not; this caused gcc 4.1 to
+	  report that it supports 'weakref', but it doesn't actually
+	  support it in the way that is needed for our optional API
+	  mechanism. The new configure script test will properly
+	  distinguish between full support and partial support for this
+	  attribute, among others. ........
+
+	* /, CHANGES: Merged revisions 200726 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 |
+	  kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6
+	  lines Document the new automatic 'ignoresdpversion' behavior.
+	  Asterisk will now automatically ignore incorrect incoming SDP
+	  version numbers when necessary to complete a T.38 re-INVITE
+	  operation. ........
+
+	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+	  165180,200689 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 |
+	  mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14
+	  lines This patch adds a new 'ignoresdpversion' option to
+	  sip.conf. When this is enabled (either globally or for a specific
+	  peer), chan_sip will treat any SDP data it receives as new data
+	  and update the media stream accordingly. By default, Asterisk
+	  will only modify the media stream if the SDP session version
+	  received is different from the current SDP session version. This
+	  option is required to interoperate with devices that have
+	  non-standard SDP session version implementations (observed by toc
+	  on the bug tracker with Microsoft OCS which always uses 0 as the
+	  session version). http://reviewboard.digium.com/r/94/ (closes
+	  issue #13958) Reported by: toc Tested by: toc ........ r200689 |
+	  kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12
+	  lines Accept T.38 re-INVITE responses with invalid SDP versions.
+	  This commit changes the 'incoming SDP version' check logic a bit
+	  more; when 'ignoresdpversion' is *not* set for a peer, if we
+	  initiate a re-INVITE to switch to T.38, we'll always accept the
+	  peer's SDP response, even if they don't properly increment the
+	  SDP version number as they should. If this situation occurs, a
+	  warning message will be generated suggesting that the peer's
+	  configuration be changed to include the 'ignoresdpversion'
+	  configuration option (although ideally they'd fix their SIP
+	  implementation to be RFC compliant). AST-221 ........
+
+2009-06-15 15:22 +0000 [r200515]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun
+	  2009) | 11 lines Merged revisions 200513 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
+	  2009) | 5 lines Add INFO to our allowed methods so that endpoints
+	  know they may send it to us. AST-223 ........ ................
+
+2009-06-12 19:08 +0000 [r200362]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c, /: Merged revisions 200361 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun
+	  2009) | 16 lines Merged revisions 200360 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
+	  2009) | 10 lines Suppress a warning message and give a better
+	  return code when generating inband ringing after a call is
+	  answered. (closes issue #15158) Reported by: madkins Patches:
+	  15158.patch uploaded by mmichelson (license 60) Tested by:
+	  madkins ........ ................
+
+2009-06-11 22:42 +0000 [r200228]  Sean Bright <sean at malleable.com>
+
+	* Makefile, /: Merged revisions 199781 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 |
+	  seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2
+	  lines Fix all of the parallel build warnings issued when running
+	  make -j#. ........
+
+2009-06-11 21:18 +0000 [r200149]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 |
+	  mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5
+	  lines Fix a crash due to a potentially NULL p->options. Thanks to
+	  mnicholson for pointing it out. ........
+
+2009-06-11 12:16 +0000 [r200040]  Leif Madsen <lmadsen at digium.com>
+
+	* /, build_tools/make_version_c, build_tools/make_version_h: Merged
+	  revisions 200039 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
+	  lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
+	  Fix path for .flavor and .version (issue #14737) Reported by:
+	  davidw Patches: flavor.patch uploaded by davidw (license 780)
+	  Tested by: davidw ........
+
+2009-06-10 20:29 +0000 [r199994]  David Brooks <dbrooks at digium.com>
+
+	* main/pbx.c, /: Fixes the argument order in definition of
+	  new_find_extension(). In the definition of new_find_extension(),
+	  the arguments 'callerid' and 'label' were swapped. The prototype
+	  declaration and all calls to the function are ordered 'callerid'
+	  then 'label', but the function itself was ordered 'label' then
+	  'callerid'. (closes issue #15303) Reported by: JimDickenson
+
+2009-06-10 20:20 +0000 [r199975]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: The 1.6.0 branch was missing all
+	  invite_branch logic. It has now been added.
+
+2009-06-10 16:13 +0000 [r199858]  Sean Bright <sean at malleable.com>
+
+	* include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
+	  (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
+	  10 Jun 2009) | 2 lines __WORDSIZE is not available on all
+	  platforms, so use sizeof(void *) instead. ........
+	  ................
+
+2009-06-09 20:54 +0000 [r199821]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
+	  dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
+	  CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
+	  only used UDP rather than copying the transport type from the
+	  peer. (closes issue #15283) Reported by: jthurman Patches:
+	  sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
+	  Tested by: jthurman, dvossel ........
+
+2009-06-08 19:39 +0000 [r199632]  Sean Bright <sean at malleable.com>
+
+	* include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
+	  (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
+	  2009) | 21 lines Increase the size of our thread stack on 64 bit
+	  processors. We were setting the stack size for each thread to
+	  240KB regardless of architecture, which meant that in some
+	  scenarios we actually had less available stack space on 64 bit
+	  processors (pointers use 8 bytes instead of 4). So now we
+	  calculate the stack size we reserve based on the platform's
+	  __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+	  bit -> 1008KB (that's right, we're ready for 128 bit processors)
+	  Patch typed by me but written by several members of
+	  #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+	  issue #14932) Reported by: jpiszcz Patches:
+	  06052009_issue14932.patch uploaded by seanbright (license 71)
+	  Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
+	  15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
+	  stack size calculation just introduced. ........ ................
+
+2009-06-05 21:37 +0000 [r199301]  David Vossel <dvossel at digium.com>
+
+	* main/pbx.c, /: Merged revisions 199298 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
+	  | 21 lines Merged revisions 199297 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
+	  | 14 lines Fixes issue with hints giving unexpected results.
+	  Hints with two or more devices that include ONHOLD gave
+	  unexpected results. (closes issue #15057) Reported by:
+	  p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+	  (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+	  devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+	  p_lindheimer, dvossel Review:
+	  https://reviewboard.asterisk.org/r/254/ ........ ................
+
+2009-06-05 13:51 +0000 [r199228]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
+	  2009) | 14 lines Correct "dahdi show channels" output when
+	  specifying a group. Since a DAHDI channel may belong to multiple
+	  groups, we need to use a bitwise and instead of equivalence to
+	  determine whether to display the channel information. (closes
+	  issue #15248) Reported by: gentian Patches: 15248.patch uploaded
+	  by mmichelson (license 60) Tested by: gentian ........
+
+2009-06-04 19:16 +0000 [r199142]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_iax2.c: Merged revisions 199139 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
+	  (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
+	  Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
+	  ................
+
+2009-06-04 14:53 +0000 [r199052]  Sean Bright <sean at malleable.com>
+
+	* main/asterisk.c, main/loader.c, /, include/asterisk/_private.h:
+	  Merged revisions 199051 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
+	  2009) | 47 lines Merged revisions 199022 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
+	  2009) | 40 lines Safely handle AMI connections/reload requests
+	  that occur during startup. During asterisk startup, a lock on the
+	  list of modules is obtained by the primary thread while each
+	  module is initialized. Issue 13778 pointed out a problem with
+	  this approach, however. Because the AMI is loaded before other
+	  modules, it is possible for a module reload to be issued by a
+	  connected client (via Action: Command), causing a deadlock. The
+	  resolution for 13778 was to move initialization of the manager to
+	  happen after the other modules had already been lodaded. While
+	  this fixed this particular issue, it caused a problem for users
+	  (like FreePBX) who call AMI scripts via an #exec in a
+	  configuration file (See issue 15189). The solution I have come up
+	  with is to defer any reload requests that come in until after the
+	  server is fully booted. When a call comes in to ast_module_reload
+	  (from wherever) before we are fully booted, the request is added
+	  to a queue of pending requests. Once we are done booting up, we
+	  then execute these deferred requests in turn. Note that I have
+	  tried to make this a bit more intelligent in that it will not
+	  queue up more than 1 request for the same module to be reloaded,
+	  and if a general reload request comes in ('module reload') the
+	  queue is flushed and we only issue a single deferred reload for
+	  the entire system. As for how this will impact existing
+	  installations - Before 13778, a reload issued before module
+	  initialization was completed would result in a deadlock. After
+	  13778, you simply couldn't connect to the manager during startup
+	  (which causes problems with #exec-that-calls-AMI configuration
+	  files). I believe this is a good general purpose solution that
+	  won't negatively impact existing installations. (closes issue
+	  #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
+	  06032009_15189_deferred_reloads.diff uploaded by seanbright
+	  (license 71) Tested by: p_lindheimer, seanbright Review:
+	  https://reviewboard.asterisk.org/r/272/ ........ ................
+
+2009-06-03 15:27 +0000 [r198825-198889]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c, /, main/features.c, include/asterisk/channel.h:
+	  Merged revisions 198856 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
+	  dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
+	  Generic call forward api, ast_call_forward() The function
+	  ast_call_forward() forwards a call to an extension specified in
+	  an ast_channel's call_forward string. After an ast_channel is
+	  called, if the channel's call_forward string is set this function
+	  can be used to forward the call to a new channel and terminate
+	  the original one. I have included this api call in both
+	  channel.c's ast_request_and_dial() and feature.c's
+	  feature_request_and_dial(). App_dial and app_queue already
+	  contain call forward logic specific for their application and
+	  options. (closes issue #13630) Reported by: festr Review:
+	  https://reviewboard.asterisk.org/r/271/ ........
+
+	* /, channels/chan_iax2.c: Merged revisions 198824 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........

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