[asterisk-commits] bebuild: tag 10.7.0-digiumphones-rc1 r369933 - /tags/10.7.0-digiumphones-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jul 11 10:53:34 CDT 2012


Author: bebuild
Date: Wed Jul 11 10:53:29 2012
New Revision: 369933

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369933
Log:
Importing files for 10.7.0-digiumphones-rc1 release.

Added:
    tags/10.7.0-digiumphones-rc1/.lastclean   (with props)
    tags/10.7.0-digiumphones-rc1/.version   (with props)
    tags/10.7.0-digiumphones-rc1/ChangeLog   (with props)

Added: tags/10.7.0-digiumphones-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1/.lastclean?view=auto&rev=369933
==============================================================================
--- tags/10.7.0-digiumphones-rc1/.lastclean (added)
+++ tags/10.7.0-digiumphones-rc1/.lastclean Wed Jul 11 10:53:29 2012
@@ -1,0 +1,3 @@
+39
+
+

Propchange: tags/10.7.0-digiumphones-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/10.7.0-digiumphones-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/10.7.0-digiumphones-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/10.7.0-digiumphones-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1/.version?view=auto&rev=369933
==============================================================================
--- tags/10.7.0-digiumphones-rc1/.version (added)
+++ tags/10.7.0-digiumphones-rc1/.version Wed Jul 11 10:53:29 2012
@@ -1,0 +1,1 @@
+10.7.0-digiumphones-rc1

Propchange: tags/10.7.0-digiumphones-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/10.7.0-digiumphones-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/10.7.0-digiumphones-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/10.7.0-digiumphones-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1/ChangeLog?view=auto&rev=369933
==============================================================================
--- tags/10.7.0-digiumphones-rc1/ChangeLog (added)
+++ tags/10.7.0-digiumphones-rc1/ChangeLog Wed Jul 11 10:53:29 2012
@@ -1,0 +1,26385 @@
+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
+	  ................
+
+	* addons/chan_ooh323.c, /: Merged revisions 369091 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10 ........
+	  r369091 | may | 2012-06-19 18:32:06 -0500 (Tue, 19 Jun 2012) | 9
+	  lines check rtptimeouts in ooh323 channels as per config file
+	  (rtp voice, video, udptl except rtcp) (closes issue
+	  ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
+	  19179-ooh323-ast10.patch ........
+
+	* /, channels/chan_sip.c: Merged revisions 369067 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369067 | mmichelson | 2012-06-19 10:37:37 -0500
+	  (Tue, 19 Jun 2012) | 17 lines Fix request routing issue when
+	  outboundproxy is used. Asterisk was incorrectly setting the
+	  destination of CANCELs and ACKs for error responses to the URI of
+	  the initial INVITE. This resulted in further requests, such as
+	  INVITEs with authentication credentials, to be routed
+	  incorrectly. Instead, when these CANCEL or ACKs are to be sent,
+	  we should simply keep the destination the same as what it
+	  previously was. There is no need to alter it any. (closes issue
+	  ASTERISK-20008) Reported by Marcus Hunger Patches:
+	  ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
+	  ........ Merged revisions 369066 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* main/features.c, /: Merged revisions 369044 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369044 | rmudgett | 2012-06-18 13:11:30 -0500
+	  (Mon, 18 Jun 2012) | 12 lines Fix monitoring calls put in a
+	  parking lot. * Fix a regression that was introduced by -r366167
+	  which effectively disabled monitoring parked calls. (closes issue
+	  ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett
+	  ........ Merged revisions 369043 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-15 16:30 +0000 [r369026]  Jason Parker <jparker at digium.com>
+
+	* apps/app_voicemail.c, /: Fix voicemail API tests by using the
+	  correct argument order for create/destroy. ........ Merged
+	  revisions 369024 from
+	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-06-15 16:25 +0000 [r369023]  Automerge script <automerge at asterisk.org>
+
+	* main/translate.c, channels/vcodecs.c,
+	  channels/sip/security_events.c, main/jitterbuf.c,
+	  main/autochan.c, pbx/dundi-parser.c, main/aoc.c, main/cel.c,
+	  main/enum.c, channels/iax2-parser.c, main/fskmodem.c,
+	  main/config.c, channels/misdn_config.c, main/netsock.c,
+	  build_tools/find_missing_support_level (added), main/loader.c,
+	  main/ulaw.c, main/dial.c, channels/sig_analog.c, main/srv.c,
+	  main/heap.c, main/privacy.c, channels/misdn/ie.c, res/ais/evt.c,
+	  main/syslog.c, res/snmp/agent.c, main/event.c, main/astmm.c,
+	  channels/sip/config_parser.c, channels/vgrabbers.c, main/db.c,
+	  main/udptl.c, main/lock.c, channels/sip/sdp_crypto.c,
+	  main/stun.c, main/frame.c, channels/sip/srtp.c,
+	  main/threadstorage.c, channels/console_video.c,
+	  channels/iax2-provision.c, main/xml.c, main/astfd.c,
+	  main/taskprocessor.c, utils/astdb2bdb.c,
+	  apps/confbridge/conf_config_parser.c, main/channel.c, main/cdr.c,
+	  res/ael/pval.c, channels/chan_misdn.c, main/framehook.c,
+	  main/tdd.c, main/strcompat.c, channels/console_gui.c,
+	  channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
+	  main/callerid.c, main/file.c, main/app.c,
+	  main/stdtime/localtime.c, main/dns.c, main/message.c,
+	  main/datastore.c, main/sched.c, main/timing.c, main/netsock2.c,
+	  main/fskmodem_float.c, /, main/slinfactory.c, main/acl.c,
+	  channels/sip/reqresp_parser.c, channels/sig_pri.c,
+	  channels/misdn/isdn_lib.c, main/term.c, main/io.c,
+	  main/hashtab.c, main/format_cap.c, main/abstract_jb.c,
+	  main/fskmodem_int.c, main/logger.c, main/audiohook.c,
+	  main/bridging.c, main/dsp.c, main/global_datastores.c,
+	  main/autoservice.c, main/dnsmgr.c, main/security_events.c,
+	  main/say.c, main/utils.c, channels/misdn/isdn_msg_parser.c,
+	  utils/astdb2sqlite3.c, main/devicestate.c, main/ssl.c,
+	  main/format_pref.c, main/astobj2.c, main/indications.c,
+	  main/chanvars.c, main/cli.c, main/tcptls.c, main/data.c,
+	  main/plc.c, main/test.c, channels/console_board.c,
+	  channels/misdn/portinfo.c, main/image.c, main/alaw.c,
+	  channels/sig_ss7.c, main/asterisk.c, main/xmldoc.c,
+	  main/format.c, main/strings.c, main/pbx.c, main/rtp_engine.c,
+	  main/ccss.c, res/ais/clm.c: Merged revisions 369005 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r369005 | kpfleming | 2012-06-15 11:07:08 -0500
+	  (Fri, 15 Jun 2012) | 22 lines Multiple revisions 369001-369002
+	  ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15
+	  Jun 2012) | 11 lines Add support-level indications to many more
+	  source files. Since we now have tools that scan through the
+	  source tree looking for files with specific support levels, we
+	  need to ensure that every file that is a component of a 'core' or
+	  'extended' module (or the main Asterisk binary) is explicitly
+	  marked with its support level. This patch adds support-level
+	  indications to many more source files in tree, but avoids adding
+	  them to third-party libraries that are included in the tree and
+	  to source files that don't end up involved in Asterisk itself.
+	  ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15
+	  Jun 2012) | 3 lines Add a script to enable finding source files
+	  without support-levels defined. ........ Merged revisions
+	  369001-369002 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-15 15:32 +0000 [r368963-368999]  Jason Parker <jparker at digium.com>
+
+	* apps/app_voicemail.exports.in, /: Remove some symbol exports that
+	  got missed in the removal of global symbols. (issue AST-807)
+	  (issue AST-901) (issue AST-908) ........ Merged revisions 368998
+	  from
+	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+	* apps/app_voicemail.c, /: These functions that were moved need to
+	  be static. Also wrap test functions in a #ifdef. (issue AST-807)
+	  (issue AST-901) (issue AST-908) ........ Merged revisions 368964
+	  from
+	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+	* tests/test_voicemail_api.c, main/app.c,
+	  include/asterisk/app_voicemail.h, apps/app_voicemail.c,
+	  include/asterisk/app.h, /: Remove global symbol requirement from
+	  app_voicemail. This uses the existing "function installation"
+	  stuff that already existed for other functions, like getting
+	  message counts. (closes issue AST-807) (issue AST-901) (issue
+	  AST-908) Review: https://reviewboard.asterisk.org/r/1965/
+	  ........ Merged revisions 368962 from
+	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-06-14 18:20 +0000 [r368872-368960]  Automerge script <automerge at asterisk.org>
+
+	* /, channels/chan_skinny.c: Merged revisions 368947 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10 ........
+	  r368947 | mjordan | 2012-06-14 12:31:33 -0500 (Thu, 14 Jun 2012)
+	  | 21 lines AST-2012-009: Fix crash in chan_skinny due to Key Pad
+	  Button Message handling AST-2012-008 (r367844) fixed a denial of
+	  service attack exploitable in the Skinny channel driver that
+	  occurred when certain messages are sent after a previously
+	  registered station sends an Off Hook message. Unresolved in that
+	  patch is an issue in the Asterisk 10 releases, wherein, if a
+	  Station Key Pad Button Message is processed after an Off Hook
+	  message, the channel driver will inappropriately dereference a
+	  NULL pointer. This patch fixes those places where the message
+	  handling or the channel callback functions would attempt to
+	  dereference the line's pointer to the device. (issue
+	  ASTERISK-19905) Reported by: Christoph Hebeisen Tested by:
+	  mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff
+	  uploaded by mjordan (license 6283) ........
+
+	* /, main/Makefile: Merged revisions 368928 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r368928 | mmichelson | 2012-06-14 10:25:23 -0500
+	  (Thu, 14 Jun 2012) | 10 lines Revert Makefile change to remove
+	  embedding res_adsi.so The change has resulted in a linking error
+	  for certain versions of GCC. This is much worse than the original
+	  issue, so for now, temporarily revert the change. A more thorough
+	  change will be sought out. ........ Merged revisions 368927 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* res/res_adsi.c, res/res_smdi.c, /, funcs/func_volume.c: Merged
+	  revisions 368895,368899 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r368895 | mjordan | 2012-06-13 15:27:28 -0500
+	  (Wed, 13 Jun 2012) | 21 lines Mark res_smdi/res_adsi as 'core'
+	  supported modules Recently, various issues surrounding weak
+	  attributes have caused problems with modules that rely on that
+	  feature to be enabled in menuselect. This includes app_voicemail
+	  and chan_dahdi, as they both rely upon res_smdi and res_adsi,
+	  which, in certain circumstances, may not be enabled by default in
+	  menuselect. Because res_smdi/res_adsi are dependencies for
+	  chan_dahdi/app_voicemail, this patch marks both as 'core'
+	  supported modules. This will allow both app_voicemail and
+	  chan_dahdi to be enabled as well, regardless of whether or not
+	  that system supports weak attributes. (issue AST-900) Reported
+	  by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto
+	  Martinez ........ Merged revisions 368894 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r368899 | mmichelson | 2012-06-13 16:13:30 -0500
+	  (Wed, 13 Jun 2012) | 19 lines Fix a deadlock that occurs when
+	  func_volume is used on a local channel. This was discovered by
+	  trying to perform a call forward to an extension that makes use
+	  of func_volume. When the local channel is optimized away, the
+	  datastore on the local;2 channel would have its audiohook
+	  destroyed rather than detaching the audiohook from the channel
+	  and then destroying it. With this patch, func_volume's datastore
+	  destructor takes the proper route of detaching the audiohook and
+	  then destroying it. (closes issue ASTERISK-19611) reported by
+	  Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
+	  Michelson (license #5049) ........ Merged revisions 368898 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, main/Makefile: Merged revisions 368885 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r368885 | mmichelson | 2012-06-13 14:36:39 -0500
+	  (Wed, 13 Jun 2012) | 11 lines Remove forced linking of res_adsi.o
+	  In GCC 4.5+ the result is that Asterisk has a phantom module
+	  loaded at startup, claiming to be res_adsi. (closes issue
+	  ASTERISK-19920) reported by Leif Madsen ........ Merged revisions
+	  368873 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* Makefile, /: Merged revisions 368831,368853 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r368831 | mjordan | 2012-06-12 13:30:06 -0500
+	  (Tue, 12 Jun 2012) | 24 lines Do not perform install on existing
+	  directories If a directory already exists, performing a 'make
+	  install' will remove the permissions associated with the current
+	  directory and replace them with the permissions of the user
+	  executing the install. This patch changes this behavior to only
+	  perform an install on the directory if the directory does not
+	  exist. Thus, if a user later changes the permissions on that
+	  directory, those permissions will be preserved in subsequent
+	  installs. Review: https://reviewboard.asterisk.org/r/1986 Review:
+	  https://reviewboard.asterisk.org/r/1864 (closes issue
+	  ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
+	  Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
+	  by mjordan) ........ Merged revisions 368830 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r368853 | mjordan | 2012-06-13 09:30:34 -0500
+	  (Wed, 13 Jun 2012) | 11 lines Do not install empty directories;
+	  add ASTLIBDIR r368830 modified the installation script to only
+	  create a directory if that directory does not exist. If some
+	  directory variable was empty, it would attempt to create the
+	  empty location. It also failed to create the ASTLIBDIR directory.
+	  This patch fixes it such that the correct directories are made
+	  and only created if a value specifying them actually exists.
+	  ........ Merged revisions 368852 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-06-12 16:22 +0000 [r368810-368826]  Jason Parker <jparker at digium.com>
+
+	* /: Let's fix the 1.8-merged prop, to give automerge the best
+	  chance at succeeding.
+
+	* funcs/func_strings.c, channels/sip/reqresp_parser.c,
+	  include/asterisk/channel.h, apps/app_queue.c,
+	  channels/chan_iax2.c, main/loader.c, main/channel.c,
+	  channels/chan_dahdi.c, channels/sig_analog.c,
+	  res/res_config_odbc.c, channels/sip/dialplan_functions.c,
+	  apps/app_directory.c, pbx/pbx_config.c, main/md5.c,
+	  res/res_odbc.c, res/res_speech.c, apps/app_voicemail.c,
+	  main/udptl.c, channels/sip/sdp_crypto.c, channels/chan_sip.c, /,
+	  res/res_fax.c, main/say.c: Multiple revisions
+	  368721,368739,368760,368808 ........ r368721 | kmoore |
+	  2012-06-11 09:11:14 -0500 (Mon, 11 Jun 2012) | 8 lines Fix
+	  compilation in dev-mode Backport a compilation fix in md5.c from
+	  trunk that only showed up in dev-mode under certain compiler
+	  versions. ........ Merged revisions 368719 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r368739 | kmoore | 2012-06-11 10:15:07 -0500 (Mon, 11 Jun 2012) |
+	  10 lines Fix coverity UNUSED_VALUE findings in core support level
+	  files Most of these were just saving returned values without
+	  using them and in some cases the variable being saved to could be
+	  removed as well. (issue ASTERISK-19672) ........ Merged revisions
+	  368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ r368760 | rmudgett | 2012-06-11 12:08:50 -0500 (Mon, 11
+	  Jun 2012) | 17 lines Fix deadlock potential with
+	  ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
+	  the channel lock held can result in a deadlock because the
+	  function also locks the bridged channel. (issue ASTERISK-19537)
+	  (closes issue AST-891) Reported by: Guenther Kelleter Tested by:
+	  Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
+	  Davis ........ Merged revisions 368759 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r368808 | mmichelson | 2012-06-12 10:37:38 -0500 (Tue, 12 Jun
+	  2012) | 15 lines Set the Caller ID "tag" on peers even if remote
+	  party information is present. On incoming calls, we were setting
+	  the cid_tag on the dialog only if there was no remote party

[... 25720 lines stripped ...]



More information about the asterisk-commits mailing list