[asterisk-commits] bebuild: tag 10.8.0-digiumphones-rc1 r370615 - /tags/10.8.0-digiumphones-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jul 31 08:35:23 CDT 2012


Author: bebuild
Date: Tue Jul 31 08:35:19 2012
New Revision: 370615

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370615
Log:
Importing files for 10.8.0-digiumphones-rc1 release.

Added:
    tags/10.8.0-digiumphones-rc1/.lastclean   (with props)
    tags/10.8.0-digiumphones-rc1/.version   (with props)
    tags/10.8.0-digiumphones-rc1/ChangeLog   (with props)

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--- tags/10.8.0-digiumphones-rc1/ChangeLog (added)
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@@ -1,0 +1,26671 @@
+2012-07-31  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.8.0-digiumphones-rc1 Released.
+
+2012-07-30 17:24 +0000 [r370555-370584]  Automerge script <automerge at asterisk.org>
+
+	* channels/chan_misdn.c, /: Merged revisions 370564 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370564 | rmudgett | 2012-07-30 11:49:12 -0500
+	  (Mon, 30 Jul 2012) | 5 lines Release B channel allocation on
+	  error path in chan_misdn. ........ Merged revisions 370563 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, apps/app_meetme.c: Merged revisions 370547 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10 ........
+	  r370547 | jrose | 2012-07-30 09:50:34 -0500 (Mon, 30 Jul 2012) |
+	  5 lines app_meetme: Change app_meetme support level to extended
+	  from deprecated (closes issue ASTERISK-20134) Reported by: Leif
+	  Madsen ........
+
+2012-07-25 21:22 +0000 [r370509]  Automerge script <automerge at asterisk.org>
+
+	* res/res_agi.c, /: Merged revisions 370495 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370495 | jrose | 2012-07-25 16:12:50 -0500
+	  (Wed, 25 Jul 2012) | 14 lines res_agi: Add message indicating
+	  need for \n character in verbose message The while loop
+	  responsible for reading AGI messages from a fastAGI service can
+	  end up looping indefinitely when an AGI script fails to indicate
+	  the end of a message with a \n character. This patch adds an
+	  indication that we are expecting a \n character to end the
+	  message to make it more clear to users that this is necessary if
+	  they are receiving this warning over and over. (issue
+	  ASTERISK-20061) Reported by: Eike Kuiper ........ Merged
+	  revisions 370494 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-25 03:45 +0000 [r370473]  Terry Wilson <twilson at digium.com>
+
+	* main/pbx.c, /: Revert a change that broke compilation 1) There is
+	  no such function as ast_ref() 2) The patch was originally
+	  credited as the one uploaded by Guenther Kelleter (license 6372)
+	  via issue AST-921, but the patch committed was not the patch
+	  referenced on the issue. 3) Guenther Kelleter's patch was
+	  actually correct. It moved the ast_free above the
+	  presencechange_cleanup label. I am not committing his change as
+	  it is not technically necesary--calling ast_free(NULL) is
+	  perfectly safe and I worry that moving the ast_free outside of
+	  the label could lead to future bugs if someone ever adds another
+	  failure conditional and expects 'goto presencechange_cleanup;' to
+	  clean up after everything.
+
+2012-07-24 21:08 +0000 [r370465]  Jonathan Rose <jrose at digium.com>
+
+	* main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c
+	  handle_presencechange (closes issue AST-921) Reported by:
+	  Guenther Kelleter Patches: nullptr.patch uploaded by Guenther
+	  Kelleter (license 6372)
+
+2012-07-24 17:24 +0000 [r370381-370452]  Automerge script <automerge at asterisk.org>
+
+	* channels/chan_oss.c, main/frame.c, /: Merged revisions
+	  370430,370432 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370430 | kpfleming | 2012-07-24 11:54:01 -0500
+	  (Tue, 24 Jul 2012) | 5 lines Rewrite a comment that didn't
+	  adequately explain the code it was documenting. ........ Merged
+	  revisions 370429 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r370432 | tzafrir | 2012-07-24 12:08:40 -0500
+	  (Tue, 24 Jul 2012) | 4 lines chan_oss: fix "sample rate" error
+	  message Merged revisions 370428 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, funcs/func_shell.c: Merged revisions 370384 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370384 | kpfleming | 2012-07-23 16:09:53 -0500
+	  (Mon, 23 Jul 2012) | 5 lines Improve documentation for the
+	  SHELL() dialplan function. ........ Merged revisions 370383 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* main/channel.c, /: Merged revisions 370361 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370361 | kpfleming | 2012-07-23 09:51:21 -0500
+	  (Mon, 23 Jul 2012) | 13 lines Free any datastores attached to
+	  dummy channels. Revision 370205 added the use of a datastore
+	  attached to a dummy channel to resolve a memory leak, but
+	  ast_dummy_channel_destructor() in this branch did not free
+	  datastores, resulting in a continued (but slightly smaller)
+	  memory leak. This patch backports the change to free said
+	  datastores from the Asterisk trunk. (related to issue AST-916)
+	  ........ Merged revisions 370360 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-19 22:24 +0000 [r370297]  Automerge script <automerge at asterisk.org>
+
+	* main/cel.c, res/res_rtp_asterisk.c, /: Merged revisions
+	  370271,370274,370277 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370271 | mjordan | 2012-07-19 16:37:09 -0500
+	  (Thu, 19 Jul 2012) | 49 lines Handle extremely out of order RFC
+	  2833 DTMF The current implementation of RFC 2833 DTMF handling in
+	  res_rtp_asterisk will, if a packet arrives out of order, drop the
+	  packet. This is to prevent duplicate ton generation in the
+	  Asterisk core. Since the RTP layer does not buffer data itself,
+	  this is the only option the RTP layer currently has for handling
+	  packets that arrive out of order. For the most part, this doesn't
+	  matter. For a particular digit, so long as a BEGIN packet arrives
+	  before the first END packet, the digit will be produced. If
+	  subsequent BEGIN packets arrive interleaved with the ENDs, they
+	  will be dropped; likewise, if the BEGIN or END packets themselves
+	  are out of order, those packets are dropped but sufficient
+	  information is conveyed to the Asterisk core to produce the
+	  appropriate digit. For certain sequences of DTMF packets - most
+	  notably when, for a particular digit, an END packet arrives
+	  before any BEGIN packet for that digit - this is a real problem.
+	  When an END arrives before any BEGINs, the END packet is dropped
+	  - but at the same time, it causes subsequent BEGIN packets for
+	  that digit to be ignored. When the next in order END packet
+	  arrives, it too is dropped - Asterisk believes that there was no
+	  initial BEGIN. The solution this patch provides is to trust the
+	  END packet to convey the information needed for the Asterisk core
+	  to produce the DTMF digit. If we receive an END packet, and it: *
+	  Has a timestamp greater then the last timestamp received from an
+	  END packet * Does not have the same sequence number as the last
+	  received sequence number (and is thus not an END packet
+	  retransmission) Then we send the END frame up to the Asterisk
+	  core. It contains enough DTMF information for Asterisk to produce
+	  the digit. On the other hand, if we receive a BEGIN or
+	  continuation packet that occurs with a timestamp equal to or less
+	  then the last END timestamp, then we've received something out of
+	  order - but we already have received enough information to
+	  produce the digit. These packets are dropped. Much thanks goes to
+	  Olle Johansson (oej) for providing the idea for this solution.
+	  Review: https://reviewboard.asterisk.org/r/2033/ (issue
+	  ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt
+	  Jordan ........ Merged revisions 370252 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r370274 | mjordan | 2012-07-19 17:01:32 -0500
+	  (Thu, 19 Jul 2012) | 17 lines Fix compilation error when
+	  MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel
+	  datastore was introduced whose destruction function pointer was
+	  pointed to the ast_free macro. Without MALLOC_DEBUG enabled this
+	  compiles as fine, as ast_free is defined as free. With
+	  MALLOC_DEBUG enabled, however, ast_free takes on a definition
+	  from a different place then utils.h, and became undefined. This
+	  patch resolves this by using a reference to ast_free_ptr. When
+	  MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG
+	  is not enabled, this is defined to be ast_free, which is defined
+	  to be free. (issue AST-916) Reported by: Thomas Arimont ........
+	  Merged revisions 370273 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r370277 | rmudgett | 2012-07-19 17:11:48 -0500
+	  (Thu, 19 Jul 2012) | 7 lines Fix compiler warnings. gcc (GCC)
+	  4.2.4 has problems casting away constness. ........ Merged
+	  revisions 370275 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-18 19:23 +0000 [r370202-370224]  Automerge script <automerge at asterisk.org>
+
+	* main/cel.c, /: Merged revisions 370206 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370206 | kpfleming | 2012-07-18 14:14:09 -0500
+	  (Wed, 18 Jul 2012) | 19 lines Resolve severe memory leak in CEL
+	  logging modules. A customer reported a significant memory leak
+	  using Asterisk 1.8. They have tracked it down to
+	  ast_cel_fabricate_channel_from_event() in main/cel.c, which is
+	  called by both in-tree CEL logging modules (cel_custom.c and
+	  cel_sqlite3_custom.c) for each and every CEL event that they log.
+	  The cause was an incorrect assumption about how data attached to
+	  an ast_channel would be handled when the channel is destroyed;
+	  the data is now stored in a datastore attached to the channel,
+	  which is destroyed along with the channel at the proper time.
+	  (closes issue AST-916) Review:
+	  https://reviewboard.asterisk.org/r/2053/ ........ Merged
+	  revisions 370205 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c,
+	  apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c,
+	  res/res_odbc.c, main/channel.c, addons/app_mysql.c, main/pbx.c,
+	  funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c: Merged
+	  revisions 370184 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370184 | kpfleming | 2012-07-18 12:13:07 -0500
+	  (Wed, 18 Jul 2012) | 10 lines Ensure that all ast_datastore_info
+	  structures are 'const'. While addressing a bug, I came across a
+	  instance of 'struct ast_datastore_info' that was not declared
+	  'const'. Since the API already expects them to be 'const', this
+	  patch changes the declarations of all existing instances that
+	  were not already declared that way. ........ Merged revisions
+	  370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-16 20:24 +0000 [r370101-370151]  Automerge script <automerge at asterisk.org>
+
+	* /, channels/chan_sip.c: Merged revisions 370132 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370132 | wdoekes | 2012-07-16 14:52:45 -0500
+	  (Mon, 16 Jul 2012) | 11 lines Code cleanup and bugfix in chan_sip
+	  outboundproxy parsing. The bug was clearing the global
+	  outboundproxy when a peer-specific outboundproxy was bad. The
+	  cleanup reduces duplicate code. Review:
+	  https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
+	  Michelson ........ Merged revisions 370131 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* UPGRADE.txt, CHANGES, UPGRADE-1.8.txt, /: Merged revisions 370082
+	  via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370082 | kmoore | 2012-07-16 08:51:57 -0500
+	  (Mon, 16 Jul 2012) | 8 lines Add comments about the BUILD_NATIVE
+	  change This is a significant change and mention of it should have
+	  gone into UPGRADE.txt and CHANGES. ........ Merged revisions
+	  370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-12 20:24 +0000 [r369958-370036]  Automerge script <automerge at asterisk.org>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c, /,
+	  channels/chan_sip.c: Merged revisions 370015,370025 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r370015 | kmoore | 2012-07-12 15:05:45 -0500
+	  (Thu, 12 Jul 2012) | 11 lines Include Expires header for SIP
+	  PUBLISH requests RFC3903 requres SIP PUBLISH requests to have
+	  Expires headers, so add them. Review:
+	  https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
+	  ........ Merged revisions 370014 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r370025 | rmudgett | 2012-07-12 15:20:02 -0500
+	  (Thu, 12 Jul 2012) | 8 lines Add missing ast_hangup() calls on
+	  some analog exception paths. Make starting analog_ss_thread() or
+	  __analog_ss_thread() failure paths hangup the channel. ........
+	  Merged revisions 370017 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 369994 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369994 | kmoore | 2012-07-12 13:55:17 -0500
+	  (Thu, 12 Jul 2012) | 12 lines Prevent double uri_escaping in
+	  chan_sip when pedantic is enabled If pedantic mode is enabled,
+	  outbound invites will have double-escaped contacts. This avoids
+	  setting an already-escaped string into a field where it is
+	  expected to be unescaped. (closes issue ASTERISK-20023)
+	  Reported-by: Walter Doekes ........ Merged revisions 369993 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* funcs/func_math.c, /: Merged revisions 369971 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369971 | elguero | 2012-07-12 09:25:45 -0500
+	  (Thu, 12 Jul 2012) | 14 lines Correct Documentation For DEC
+	  Function The documentation for DEC in func_math.c was incorrect.
+	  Looks like a copy and paste error. (Closes issue ASTERISK-20095)
+	  Reported by: Billy Chia Tested by: Michael L. Young Patches:
+	  func_math.patch uploaded by Billy Chia (license 6381) ........
+	  Merged revisions 369970 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* funcs/func_realtime.c, /: Merged revisions 369938 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369938 | tilghman | 2012-07-11 12:12:28 -0500
+	  (Wed, 11 Jul 2012) | 11 lines Allow the REALTIME() function to
+	  report errors back to the caller. Also, do more error checking on
+	  the arguments specified to the REALTIME() function and clarify
+	  the documentation. While I was editing the file, a few coding
+	  guidelines fixups, as well. Review:
+	  https://reviewboard.asterisk.org/r/2031/ ........ Merged
+	  revisions 369937 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-30  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.7.0-digiumphones Released.
+
+2012-07-11  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.7.0-digiumphones-rc1 Released.
+
+2012-07-10 14:22 +0000 [r369889]  Automerge script <automerge at asterisk.org>
+
+	* apps/app_stack.c, main/pbx.c, /: Merged revisions 369871 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369871 | kmoore | 2012-07-10 08:35:30 -0500
+	  (Tue, 10 Jul 2012) | 12 lines Improve Goto and GotoIf related
+	  documentation Correct documentation on labeliftrue and
+	  labeliffalse parameters of GotoIf() and update several other
+	  locations that use the same syntax. (closes issue ASTERISK-20007)
+	  Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged
+	  revisions 369869 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-09 19:51 +0000 [r369846]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Add support for exposing the received
+	  contact URI and also for setting the request URI in messages.
+	  (closes issue AST-911)
+
+2012-07-09 17:22 +0000 [r369810-369836]  Automerge script <automerge at asterisk.org>
+
+	* configs/sip_notify.conf.sample, /: Merged revisions 369819 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369819 | qwell | 2012-07-09 12:06:40 -0500
+	  (Mon, 09 Jul 2012) | 9 lines Add Digium phones context to
+	  sip_notify sample config. This makes it so that they can be
+	  reconfigured remotely. (closes issue ASTERISK-19910) ........
+	  Merged revisions 369818 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 369793 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369793 | jrose | 2012-07-09 09:43:49 -0500
+	  (Mon, 09 Jul 2012) | 9 lines chan_sip: Fix small behavioral
+	  change accidentally introduced in r369750 When removing the
+	  warning for AST_CONTROL_FLASH from sip_indicate, I also
+	  inadvertently changed the return value, which would likely make
+	  the indication not be sent in audio. This fixes that while still
+	  removing the warning message. ........ Merged revisions 369792
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-06 21:21 +0000 [r369643-369763]  Automerge script <automerge at asterisk.org>
+
+	* /, channels/chan_sip.c: Merged revisions 369751 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369751 | jrose | 2012-07-06 16:02:37 -0500
+	  (Fri, 06 Jul 2012) | 12 lines chan_sip: Add case for FLASH
+	  control frames so that we don't display a warning. chan_sip
+	  channels can receive flash control frames when connected to
+	  analog phones and possibly for other reasons. There really isn't
+	  a reason to warn when these frames are received, we can safely
+	  ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan
+	  Rose (license 6182) ........ Merged revisions 369750 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* main/tcptls.c, /: Merged revisions 369732 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369732 | mmichelson | 2012-07-06 13:47:05 -0500
+	  (Fri, 06 Jul 2012) | 21 lines Remove a superfluous and dangerous
+	  freeing of an SSL_CTX. The problem here is that multiple server
+	  sessions share a SSL_CTX. When one session ended, the SSL_CTX
+	  would be freed and set NULL, leaving the other sessions unable to
+	  function. The code being removed is superfluous because the
+	  SSL_CTX structures for servers will be properly freed when
+	  ast_ssl_teardown is called. (closes issue ASTERISK-20074)
+	  Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded
+	  by Mark Michelson (license #5049) Testers: Trevor Helmsley
+	  ........ Merged revisions 369731 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* main/bridging.c, /: Merged revisions 369709 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369709 | mmichelson | 2012-07-06 10:23:28 -0500
+	  (Fri, 06 Jul 2012) | 14 lines Fix bridging thread leak. The
+	  bridge thread was exiting but was never being reaped using
+	  pthread_join(). This has been fixed now by calling pthread_join()
+	  in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported
+	  by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012
+	  ........ Merged revisions 369708 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* apps/app_voicemail.c, /: Merged revisions 369653 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369653 | kmoore | 2012-07-05 14:12:33 -0500
+	  (Thu, 05 Jul 2012) | 20 lines Resolve heap corruption issue with
+	  voicemail The heard and deleted arrays in the voicemail state
+	  structure were not handled properly following the memory leak fix
+	  in r354890 and a fix for an invalid free in r356797. This could
+	  result in accessing and writing into freed memory. The allocation
+	  for these arrays has been reworked to avoid the possibility of
+	  invalid frees, access of freed memory, and crashes that were
+	  occurring as a result of this. Locking around accesses and
+	  modifications of the voicemail state structure members
+	  dh_arraysize, heard, and deleted has been added to prevent
+	  simultaneous modification and access when IMAP storage is in use.
+	  If IMAP storage is not in use, this locking is not compiled in.
+	  Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
+	  ASTERISK-19923) ........ Merged revisions 369652 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 369627 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369627 | mjordan | 2012-07-05 12:02:53 -0500
+	  (Thu, 05 Jul 2012) | 18 lines Do not send a BYE when a
+	  provisional response arrives during a re-INVITE Commits r369557
+	  and r369579 were done to improve handling of re-INVITEs when the
+	  UA that was supposed to receive the re-INVITE fails to respond. A
+	  limitation of those patches occurred when a UA sent a provisional
+	  response to the re-INVITE. This triggered a sending of a BYE in
+	  check_pending. This patch tweaks the handling of the re-INVITE
+	  such that a BYE is not sent in response to those messages. (issue
+	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
+	  patches: (reinvite_tweak.diff license #5012 by Steve Davies)
+	  ........ Merged revisions 369626 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-07-03 17:23 +0000 [r369578-369598]  Automerge script <automerge at asterisk.org>
+
+	* /, channels/chan_sip.c: Merged revisions 369580 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369580 | twilson | 2012-07-03 12:02:18 -0500
+	  (Tue, 03 Jul 2012) | 11 lines More improvements to re-INVITEs
+	  timing out after a provisional response There is no need to call
+	  check_pendings() on a final response to an INVITE when destroying
+	  the scheduler entry as it will be done later during normal
+	  processing. (issue ASTERISK-19992) ........ Merged revisions
+	  369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+	  revisions 369558 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369558 | twilson | 2012-07-03 09:34:22 -0500
+	  (Tue, 03 Jul 2012) | 14 lines Better handle re-INVITEs with
+	  provisional but no final repsonses A previous attempt at fixing
+	  this issue had negative side effects related to attended
+	  transfers which this patch should resolve. Many thanks to Steve
+	  Davies for all of the good suggestions and testing. (closes issue
+	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
+	  Davies, Terry Wilson Review:
+	  https://reviewboard.asterisk.org/r/2009/ ........ Merged
+	  revisions 369557 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-29 21:19 +0000 [r369488-369516]  Automerge script <automerge at asterisk.org>
+
+	* main/rtp_engine.c, /: Merged revisions 369511 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10 ........
+	  r369511 | mmichelson | 2012-06-29 15:28:10 -0500 (Fri, 29 Jun
+	  2012) | 3 lines Fix apparent copy and paste error where incorrect
+	  "glue" is used. ........
+
+	* /, channels/chan_sip.c: Merged revisions 369491 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369491 | file | 2012-06-29 11:54:11 -0500 (Fri,
+	  29 Jun 2012) | 5 lines With some configurations a transport is
+	  not actually specified so assume UDP in these cases. ........
+	  Merged revisions 369490 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 369472 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369472 | file | 2012-06-29 10:30:47 -0500 (Fri,
+	  29 Jun 2012) | 10 lines Make the address family filter specific
+	  to the transport. (closes issue ASTERISK-16618) Reported by: Leif
+	  Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........
+	  Merged revisions 369471 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-27 21:22 +0000 [r369453]  Automerge script <automerge at asterisk.org>
+
+	* channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+	  revisions 369437 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369437 | twilson | 2012-06-27 16:10:01 -0500
+	  (Wed, 27 Jun 2012) | 16 lines Clean up after a reinvite that
+	  never gets a final response The basic problem is that if a
+	  re-INVITE is sent by Asterisk and it receives a provisional
+	  response, but no final response, then the dialog is never torn
+	  down. In addition to leaking memory, this also leaks file
+	  descriptors and will eventually lead to Asterisk no longer being
+	  able to process calls. This patch just keeps track of whether
+	  there is an outstanding re-INVITE, and if there is goes ahead and
+	  cleans up everything as though there was no outstanding reinvite.
+	  (closes issue ASTERISK-19992) ........ Merged revisions 369436
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-26 14:21 +0000 [r369322-369406]  Automerge script <automerge at asterisk.org>
+
+	* main/adsi.c, /: Merged revisions 369391 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369391 | mjordan | 2012-06-26 08:22:42 -0500
+	  (Tue, 26 Jun 2012) | 15 lines Fix crash in unloading of res_adsi
+	  module When res_adsi is unloaded, it removes the ADSI functions
+	  that it previously installed by passing a NULL adsi_funcs pointer
+	  to ast_adsi_install_funcs. This function was not checking whether
+	  or not the adsi_funcs pointer passed in was NULL before
+	  dereferencing it to check whether or not the version of the
+	  functions matches what the core was expecting it. This patch
+	  makes it so that the version is only checked if a potentially
+	  valid adsi_funcs pointer was passed in. Passing in NULL removes
+	  the installed functions, bypassing the version check. ........
+	  Merged revisions 369390 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* main/cdr.c, /: Merged revisions 369369 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369369 | mjordan | 2012-06-25 14:36:02 -0500
+	  (Mon, 25 Jun 2012) | 29 lines Fix incorrect duration reporting in
+	  CDRs created in batch mode Certain places in core/cdr.c would, if
+	  the duration value were 0, calculate the duration as being the
+	  delta between the current time and the time at which the CDR
+	  record was started. While this does not typically cause a problem
+	  in non-batch mode, this can cause an issue in batch mode where
+	  CDR records are gathered and written long after those calls have
+	  ended. In particular, this affects calls that were never
+	  answered, as those are expected to have a duration of 0. Often,
+	  this would result in CDR logs with a significant number of calls
+	  with lengthy durations, but dispositions of "BUSY". Note that
+	  this does not affect cdr_csv, as that backend does not use
+	  ast_cdr_getvar and instead directly reports the duration value.
+	  The affected core backends include cdr_apative_odbc and
+	  cdr_custom; other extended or deprecated CDR backends may
+	  potentially still directly manipulate the duration values. (issue
+	  ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
+	  Reported by: Thomas Arimont Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/1996/ ........ Merged
+	  revisions 369351 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+	  revisions 369353 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369353 | mmichelson | 2012-06-25 14:16:52 -0500
+	  (Mon, 25 Jun 2012) | 14 lines Re-fix how local tag is generated
+	  when sending a 481 to an INVITE. Match our local tag to whatever
+	  to-tag was sent in the initial INVITE. Because the size of the
+	  to-tag may not fit in the buffer in the sip_pvt, it has been
+	  changed to a string field. (closes issue ASTERISK-19892) reported
+	  by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977
+	  ........ Merged revisions 369352 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* main/features.c, res/res_adsi.c, main/adsi.c (added),
+	  res/res_adsi.exports.in (removed), include/asterisk/adsi.h, /,
+	  main/Makefile: Merged revisions 369325,369328 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369325 | mmichelson | 2012-06-25 10:52:42 -0500
+	  (Mon, 25 Jun 2012) | 20 lines Multiple revisions 369323-369324
+	  ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon,
+	  25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module.
+	  The way this is done is to stop using the optional API. Instead,
+	  res_adsi.so, when loaded fills in a table of function pointers.
+	  Review: https://reviewboard.asterisk.org/r/1991 ........ r369324
+	  | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2
+	  lines Forgot to svn add this file in my last commit. ........
+	  Merged revisions 369323-369324 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r369328 | rmudgett | 2012-06-25 10:59:28 -0500
+	  (Mon, 25 Jun 2012) | 15 lines Fix Bridge application occasionally
+	  returning to the wrong location. * Fix do_bridge_masquerade()
+	  getting the resume location from the zombie channel. The code
+	  must not touch a clone channel after it has masqueraded it. The
+	  clone channel has become a zombie and is starting to hangup.
+	  (closes issue ASTERISK-19985) Reported by: jamicque Patches:
+	  jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
+	  rmudgett Tested by: jamicque ........ Merged revisions 369327
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 369303 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369303 | mmichelson | 2012-06-25 09:23:16 -0500
+	  (Mon, 25 Jun 2012) | 14 lines Be more consistent with the return
+	  code for requests received from invalid domain. When Asterisk
+	  receives an INVITE from an external domain when
+	  allowexternaldomains=no send a 403 instead of a 404. This is
+	  consistent with Asterisk's behavior when receiving a REGISTER in
+	  this situation. (Closes issue ASTERISK-19601) Reported by Matthew
+	  Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark
+	  Michelson (License #5049) ........ Merged revisions 369302 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-23 00:20 +0000 [r369213-369294]  Automerge script <automerge at asterisk.org>
+
+	* main/features.c, /: Merged revisions 369283 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369283 | rmudgett | 2012-06-22 19:12:27 -0500
+	  (Fri, 22 Jun 2012) | 22 lines Fix Bridge application and AMI
+	  Bridge action error handling. * Fix AMI Bridge action
+	  disconnecting the AMI link on error. * Fix AMI Bridge action and
+	  Bridge application not checking if their masquerades were
+	  successful. * Fix Bridge application running the h-exten when it
+	  should not. * Made do_bridge_masquerade() return if the
+	  masquerade was successful so the Bridge application and AMI
+	  Bridge action could deal with it correctly. * Made
+	  bridge_call_thread_launch() hangup the passed in channels if the
+	  bridge_call_thread fails to start. Those channels would have been
+	  orphaned. * Made builtin_atxfer() check the success of the
+	  transfer masquerade setup. ........ Merged revisions 369282 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* apps/app_queue.c, apps/app_dial.c, /: Merged revisions
+	  369259,369263 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369259 | rmudgett | 2012-06-22 16:37:05 -0500
+	  (Fri, 22 Jun 2012) | 5 lines Check if PBX was started and fix F
+	  and F(x) action logic in Dial application. ........ Merged
+	  revisions 369258 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r369263 | rmudgett | 2012-06-22 17:09:29 -0500
+	  (Fri, 22 Jun 2012) | 5 lines Explicitly check caller hangup in
+	  app Queue rather than a polluted res2 value. ........ Merged
+	  revisions 369262 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c, main/ccss.c: Merged revisions
+	  369236,369239 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369236 | rmudgett | 2012-06-22 15:49:33 -0500
+	  (Fri, 22 Jun 2012) | 5 lines Change incorrect chan_sip zombie
+	  hangup debug message. They are all zombies now. ........ Merged
+	  revisions 369235 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r369239 | rmudgett | 2012-06-22 16:04:25 -0500
+	  (Fri, 22 Jun 2012) | 5 lines Check if PBX was started for generic
+	  CCSS recall. ........ Merged revisions 369238 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 369215 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369215 | twilson | 2012-06-22 14:34:59 -0500
+	  (Fri, 22 Jun 2012) | 9 lines Don't crash on a guest directmedia
+	  call A sip_pvt may not have relatedpeer set if a call doesn't
+	  match up with a peer. If there is no relatedpeer, there is no
+	  direct media ACL to apply, so just return that it is allowed.
+	  ........ Merged revisions 369214 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 369206 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369206 | kmoore | 2012-06-22 12:23:26 -0500
+	  (Fri, 22 Jun 2012) | 11 lines Don't parse media stream state for
+	  SIP video streams The sendonly/recvonly/sendrecv/inactive media
+	  stream attributes were parsed for video, but nothing was ever
+	  done with them. With this code removed, an UNSUPPORTED message is
+	  produced when these attributes are used in conjunction with a
+	  video stream which is the better behavior since they were never
+	  really supported in the first place. ........ Merged revisions
+	  369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-20 18:22 +0000 [r369056-369164]  Automerge script <automerge at asterisk.org>
+
+	* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, /:
+	  Merged revisions 369147 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369147 | may | 2012-06-20 12:36:27 -0500 (Wed,
+	  20 Jun 2012) | 10 lines fix locking issue on empty callList
+	  (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches:
+	  ASTERISK-18322-2.patch ........ Merged revisions 369146 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* include/asterisk/netsock2.h, main/netsock2.c, /: Merged revisions
+	  369109 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369109 | elguero | 2012-06-19 21:04:58 -0500
+	  (Tue, 19 Jun 2012) | 23 lines Fix NULL pointer segfault in
+	  ast_sockaddr_parse() While working with ast_parse_arg() to
+	  perform a validity check, a segfault occurred. The segfault
+	  occurred due to passing a NULL pointer to ast_sockaddr_parse()
+	  from ast_parse_arg(). According to the documentation in config.h,
+	  "result pointer to the result. NULL is valid here, and can be
+	  used to perform only the validity checks." This patch fixes the
+	  segfault by checking for a NULL pointer. This patch also adds
+	  documentation to netsock2.h about why it is necessary to check
+	  for a NULL pointer. (Closes issue ASTERISK-20006) Reported by:
+	  Michael L. Young Tested by: Michael L. Young Patches:
+	  asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young
+	  (license 5026) Review: https://reviewboard.asterisk.org/r/1990/
+	  ........ Merged revisions 369108 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+

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