[asterisk-commits] may: branch may/ooh323_ipv6_direct_rtp r369540 - in /team/may/ooh323_ipv6_dir...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Jul 2 16:57:34 CDT 2012


Author: may
Date: Mon Jul  2 16:56:44 2012
New Revision: 369540

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369540
Log:
Multiple revisions 368588,368606,368637,368646,368663,368668,368673-368675,368680-368681,368688,368712,368714,368722,368751,368772,368784,368793-368794,368809,368832,368854-368855,368886,368896,368900,368920-368921,368929,368948,368966,368972,368979,368985,368991,369000,369007,369013,369027-369028,369034

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  r368588 | kmoore | 2012-06-06 20:11:01 +0400 (Wed, 06 Jun 2012) | 15 lines
  
  Ensure overlapping hold flags do not conflict
  
  When changing between different modes of hold, the flags were not being
  cleared out properly causing a failure to change hold states.
  
  (closes issue ASTERISK-19919)
  Patch-by: Morten Tryfoss
  Reported-by: Morten Tryfoss
  ........
  
  Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 368587 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368606 | mjordan | 2012-06-06 21:22:11 +0400 (Wed, 06 Jun 2012) | 21 lines
  
  Add feature modifier to versions produced from branches
  
  Certain branches, such as Certified Asterisk, may have a modifier added to
  them that specifies the features available in that branch.  For branches, this
  modifier is expected to be reflected in the location of the branch in
  subversion. For example, a subversion of URL of /certified/branches/1.8.11
  would have a feature modifier of 'certified'.  This is slightly different then
  how features are determined for tags, where the feature is part of the actual
  tag name, e.g., "10.5.0-digiumphones".
  
  In keeping with the nomenclature used for tags, the feature specifier for
  branches is translated and placed after the revision numbers.  For the example
  given previously, this would result in a branch version of
  "Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
  ........
  
  Merged revisions 368604 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 368605 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368637 | mmichelson | 2012-06-06 23:25:44 +0400 (Wed, 06 Jun 2012) | 34 lines
  
  Fix a specific scenario where ACKs are not matched.
  
  If a dialog-starting INVITE contains a to-tag, then Asterisk
  will respond with a 481. In this case, the resulting incoming
  ACK would not be matched, so Asterisk would continue retransmitting
  the 481 until the transaction times out.
  
  There were two issues. Asterisk, upon creating a sip_pvt would generate
  a local tag. However, when the time came to transmit the 481, since there
  was a to-tag in the INVITE, Asterisk would place this original to-tag
  in the 481 response. When the ACK came in, Asterisk would attempt to
  match the to-tag in the ACK to the generated local tag. Unfortunately,
  Asterisk never actually transmitted a response with the generated local
  tag, so the to-tag in the ACK would not match.
  
  The other problem was that when the 481 was sent, nothing was set
  on the sip_pvt to indicate what CSeq is expected in the ACK.
  
  To fix the first problem, we zero out the to-tag seen in the incoming
  INVITE. This way, Asterisk, when time to send a response, will send
  its generated local tag instead.
  
  To fix the second problem, we set the sip_pvt's pendinginvite to the
  CSeq of the INVITE when we send a 481.
  
  (closes issue ASTERISK-19892)
  Reported by Mark Michelson
  ........
  
  Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 368629 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368646 | rmudgett | 2012-06-07 01:34:10 +0400 (Thu, 07 Jun 2012) | 20 lines
  
  Fix POTS flash hook to orignate a second call deadlock.
  
  A deadlock can occur when a POTS phone tries to flash hook to originate a
  second call for 3-way or transfer.  If another process is scanning the
  channels container when the POTS line flash hooks then a deadlock will
  occur.
  
  * Release the channel and private locks when creating a new channel as a
  result of a flash hook.
  
  (closes issue ASTERISK-19842)
  Reported by: rmudgett
  Tested by: rmudgett
  ........
  
  Merged revisions 368644 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 368645 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368663 | twilson | 2012-06-07 19:43:37 +0400 (Thu, 07 Jun 2012) | 9 lines
  
  Add default handler documentation and standardize acl handler
  
  Added documentation describing what flags and arguments to pass to
  aco_option_register for default option types. Also changed the ACL
  handler to use the flags parameter to differentiate between "permit"
  and "deny" instead of adding an additional vararg parameter.
  
  Review: https://reviewboard.asterisk.org/r/1969/
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  r368668 | tzafrir | 2012-06-08 00:00:29 +0400 (Fri, 08 Jun 2012) | 4 lines
  
  Fix a typo in format_ogg_vorbis.c: suport
  
  Review: https://reviewboard.asterisk.org/r/1970/
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  r368673 | twilson | 2012-06-08 00:32:07 +0400 (Fri, 08 Jun 2012) | 8 lines
  
  Fix reloading an unchanged file with the Config Options API
  
  Adding multiple file support broke reloading an unchanged file. This
  adds an enum for return values for the aco_process_* functions and
  ensures that the config is not applied if res is not ACO_PROCESS_OK.
  
  Review: https://reviewboard.asterisk.org/r/1979/
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  r368674 | rmudgett | 2012-06-08 00:37:05 +0400 (Fri, 08 Jun 2012) | 6 lines
  
  Fix inverted test in app_queue for ringinuse.
  
  Regression from -r367080 ringinuse commit.
  
  (issue ASTERISK-19536)
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  r368675 | rmudgett | 2012-06-08 00:39:25 +0400 (Fri, 08 Jun 2012) | 2 lines
  
  Fix app_queue debug message use of args.options after the string has been parsed.
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  r368680 | wedhorn | 2012-06-08 01:23:42 +0400 (Fri, 08 Jun 2012) | 10 lines
  
  Skinny cleanup.
  
  Removed d->registered which was mirroring d->session. Changed relevant
  references to use d->session instead.
  
  Moved setting and unsetting of l->device from session register to device 
  configuration. As such, l->device will always be valid unless it is has not
  been configured to a device. Revised various test where checking if a device
  is registered to use l->device->session.
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  r368681 | wedhorn | 2012-06-08 01:44:15 +0400 (Fri, 08 Jun 2012) | 7 lines
  
  Skinny cleanup (mwi_event_cb).
  
  Original was testing for d->session, setting and testing again (all nested).
  
  Removed duplicate testing and restructured function to test/return and then
  the main code.
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  r368688 | igorg | 2012-06-08 12:32:49 +0400 (Fri, 08 Jun 2012) | 8 lines
  
  
  Fix MWI update so LED display correct voicemail state after phone usage. Also fixes few warnings.
  (closes issue #19675)
   Reported by: dbohling
   Patches: 
         fixmwi.patch uploaded by dbohling (license 6378)
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  r368712 | rmudgett | 2012-06-09 00:49:00 +0400 (Sat, 09 Jun 2012) | 1 line
  
  Tweak ast_channel_softhangup_withcause_locked() to take a typed parameter.
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  r368714 | rmudgett | 2012-06-09 01:08:17 +0400 (Sat, 09 Jun 2012) | 7 lines
  
  Fix error paths in action_hangup() for AMI Hangup action.
  
  * Check allocation function return values for failure.  Crashing is bad.
  
  * Tweak ast_regex_string_to_regex_pattern() parameters for proper ast_str 
  usage.  
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  r368722 | kmoore | 2012-06-11 18:12:08 +0400 (Mon, 11 Jun 2012) | 11 lines
  
  Recorded merge of revisions 368721 from http://svn.asterisk.org/svn/asterisk/branches/10
  
  ........
  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
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  r368751 | kmoore | 2012-06-11 19:23:30 +0400 (Mon, 11 Jun 2012) | 13 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
  ........
  
  Merged revisions 368739 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368772 | rmudgett | 2012-06-11 21:34:08 +0400 (Mon, 11 Jun 2012) | 20 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
  ........
  
  Merged revisions 368760 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368784 | kmoore | 2012-06-12 08:03:23 +0400 (Tue, 12 Jun 2012) | 10 lines
  
  Parse ANI2 information from SIP From header parameters
  
  ANI2 information is now parsed out of SIP From headers when present in
  the oli, isup-oli, and ss7-oli parameters and is available via the
  CALLERID(ani2) dialplan function.
  
  (closes issue ASTERISK-19912)
  Patch-by: Rob Gagnon
  Review: https://reviewboard.asterisk.org/r/1947/
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  r368793 | mjordan | 2012-06-12 18:07:13 +0400 (Tue, 12 Jun 2012) | 18 lines
  
  Fix deadlock in SIP transfers that involve a REFER request
  
  In r367163, "send to voicemail" functionality was added to the SIP channel
  driver.  This required updating the party redirecting information for the
  channel based on the headers provided in the REFER request.  When the
  redirecting party information is updated on the channel, a call to
  ast_indicate_data occurs.  Because handle_request_refer still had the sip_pvt
  locked, a deadlock could occur between the pbx_thread and the do_monitor thread
  servicing the REFER request.
  
  This patch preserves the proper locking order between the channel and the
  sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party
  redirecting information on the channel.
  
  (closes issue AST-903)
  Reported by: Matt Jordan
  patches:
    jira_ast_903_trunk.patch by rmudgett (license 5621)
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  r368794 | mjordan | 2012-06-12 18:09:41 +0400 (Tue, 12 Jun 2012) | 1 line
  
  Update merge property information
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  r368809 | mmichelson | 2012-06-12 19:46:48 +0400 (Tue, 12 Jun 2012) | 18 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 information (Remote-Party-ID or P-Asserted-Identity) present.
  The Caller ID tag is an invented parameter, though, and should be set no matter
  the circumstance.
  
  (closes issue ASTERISK-19859)
  Reported by Thomas Arimont
  (closes issue AST-884)
  Reported by Trey Blancher
  ........
  
  Merged revisions 368807 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 368808 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368832 | mjordan | 2012-06-12 22:41:50 +0400 (Tue, 12 Jun 2012) | 27 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
  ........
  
  Merged revisions 368831 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368854 | mjordan | 2012-06-13 18:31:24 +0400 (Wed, 13 Jun 2012) | 14 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
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  Merged revisions 368853 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368855 | mjordan | 2012-06-13 18:55:30 +0400 (Wed, 13 Jun 2012) | 4 lines
  
  Replace MODULES_DIR with ASTMODDIR in Makefile's INSTALLDIRS
  
  Post Asterisk 10, the MODULES_DIR variable no longer exists, and was replaced
  with ASTMODDIR.
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  r368886 | mmichelson | 2012-06-13 23:51:08 +0400 (Wed, 13 Jun 2012) | 14 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
  ........
  
  Merged revisions 368885 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368896 | mjordan | 2012-06-14 00:28:07 +0400 (Thu, 14 Jun 2012) | 24 lines
  
  Mark res_smdi/res_adsi as 'core' supported modules
  
  Recently, various issues surrounding weak symbols 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 symbols.
  
  (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
  ........
  
  Merged revisions 368895 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368900 | mmichelson | 2012-06-14 01:17:13 +0400 (Thu, 14 Jun 2012) | 22 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
  ........
  
  Merged revisions 368899 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368920 | twilson | 2012-06-14 17:35:07 +0400 (Thu, 14 Jun 2012) | 8 lines
  
  Add filename alias support to the Config Options API
  
  This adds the ability to handle a single filename alias for a config
  file. This is useful if a config filename has changed, but the old
  filename should be supported for backwards compatibility.
  
  Review: https://reviewboard.asterisk.org/r/1981/ 
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  r368921 | twilson | 2012-06-14 17:41:47 +0400 (Thu, 14 Jun 2012) | 7 lines
  
  Add a post_apply callback to the Config Options API
  
  This adds a callback that only fires when changes have been successfully
  applied via the Config Options API.
  
  Review: https://reviewboard.asterisk.org/r/1980/
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  r368929 | mmichelson | 2012-06-14 19:28:02 +0400 (Thu, 14 Jun 2012) | 13 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
  ........
  
  Merged revisions 368928 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368948 | mjordan | 2012-06-14 21:34:10 +0400 (Thu, 14 Jun 2012) | 22 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)
  ........
  
  Merged revisions 368947 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r368966 | qwell | 2012-06-14 23:40:11 +0400 (Thu, 14 Jun 2012) | 35 lines
  
  Multiple revisions 368963,368965
  
  ........
    r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) | 14 lines
    
    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
  ........
    r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun 2012) | 11 lines
    
    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
  ........
  
  Merged revisions 368963,368965 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
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  r368972 | rmudgett | 2012-06-15 00:49:28 +0400 (Fri, 15 Jun 2012) | 1 line
  
  Move vm defines to group them better.
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  r368979 | rmudgett | 2012-06-15 02:57:21 +0400 (Fri, 15 Jun 2012) | 10 lines
  
  Make the Hangup application set a softhangup flag.
  
  The Hangup application used to just return -1 to cause normal dialplan
  execution to hangup a channel.  For the non-normal execution routines like
  predial and connected-line interception routines, the hangup request would
  exit the routine early but otherwise be ignored.
  
  * Made the Hangup application not allow setting a cause code of zero.  A
  zero cause code is not defined.
........
  r368985 | rmudgett | 2012-06-15 03:22:53 +0400 (Fri, 15 Jun 2012) | 27 lines
  
  Allow non-normal execution routines to be able to run on hungup channels.
  
  * Make non-normal dialplan execution routines be able to run on a hung up
  channel.  This is preparation work for hangup handler routines.
  
  * Fixed ability to support relative non-normal dialplan execution
  routines.  (i.e., The context and exten are optional for the specified
  dialplan location.) Predial routines are the only non-normal routines that
  it makes sense to optionally omit the context and exten.  Setting a hangup
  handler also needs this ability.
  
  * Fix Return application being able to restore a dialplan location
  exactly.  Channels without a PBX may not have context or exten set.
  
  * Fixes non-normal execution routines like connected line interception and
  predial leaving the dialplan execution stack unbalanced.  Errors like
  missing Return statements, popping too many stack frames using StackPop,
  or an application returning non-zero could leave the dialplan stack
  unbalanced.
  
  * Fixed the AGI gosub application so it cleans up the dialplan execution
  stack and handles the autoloop priority increments correctly.
  
  * Eliminated the need for the gosub_virtual_context return location.
  
  Review: https://reviewboard.asterisk.org/r/1984/
........
  r368991 | rmudgett | 2012-06-15 04:55:43 +0400 (Fri, 15 Jun 2012) | 1 line
  
  Remove remaining properties mmichelson left laying around from phones branch merge.
........
  r369000 | qwell | 2012-06-15 19:33:41 +0400 (Fri, 15 Jun 2012) | 12 lines
  
  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
  ........
  
  Merged revisions 368999 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
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  r369007 | kmoore | 2012-06-15 20:17:12 +0400 (Fri, 15 Jun 2012) | 11 lines
  
  Add HANGUPCAUSE hash support to IAX2
  
  Continuing with the Who Hung Up? project for Asterisk 11, this adds
  support to IAX2 for the HANGUPCAUSE hash.
  
  Additionally, this breaks out some functionality in frame.c for getting
  information about frame types and subclasses.
  
  Review: https://reviewboard.asterisk.org/r/1941/
  (issue SWP-4222)
........
  r369013 | kpfleming | 2012-06-15 20:20:16 +0400 (Fri, 15 Jun 2012) | 25 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
  ........
  
  Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r369027 | qwell | 2012-06-15 20:30:58 +0400 (Fri, 15 Jun 2012) | 8 lines
  
  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
  ........
  
  Merged revisions 369026 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
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  r369028 | kmoore | 2012-06-15 21:13:20 +0400 (Fri, 15 Jun 2012) | 16 lines
  
  Allow chan_sip to decline unwanted media streams
  
  This change replaces the static array of four representable media
  streams with an AST_LIST so that chan_sip can keep track of offered
  media streams.  This allows chan_sip to deal with offers containing
  multiple same-type streams and many other situations without rejecting
  the SDP offer in its entirety, yet still generating a valid response.
  This also covers cases where Asterisk can not comprehend the offer if
  it is in the correct format.
  
  Previously, chan_sip would reject SDP offers or entirely ignore
  individual stream offers in an effort to be more compatible which
  would often result in invalid SDP responses.
  
  Review: https://reviewboard.asterisk.org/r/1988/
........
  r369034 | wedhorn | 2012-06-16 01:18:56 +0400 (Sat, 16 Jun 2012) | 14 lines
  
  Various small chan_skinny fixes and cleanup
  
  Added test to skinny_register to only allow device to register against
  a device that is not already registered.
  
  Addback l->device test for skinny_show_lines. Fixes segfault if a line
  is configured but not configured to a device. Reverses part of r368680.
  
  Removed redundant l->device tests in subsubstate and dumpsub. l->device
  will always be valid if these routines are called. Reverses 368948 -
  discussed with mjordan on irc.
  
  Some indentation cleanup.
........

Merged revisions 368588,368606,368637,368646,368663,368668,368673-368675,368680-368681,368688,368712,368714,368722,368751,368772,368784,368793-368794,368809,368832,368854-368855,368886,368896,368900,368920-368921,368929,368948,368966,368972,368979,368985,368991,369000,369007,369013,369027-369028,369034 from http://svn.asterisk.org/svn/asterisk/trunk

Added:
    team/may/ooh323_ipv6_direct_rtp/build_tools/find_missing_support_level
      - copied unchanged from r369034, trunk/build_tools/find_missing_support_level
Removed:
    team/may/ooh323_ipv6_direct_rtp/include/asterisk/app_voicemail.h
Modified:
    team/may/ooh323_ipv6_direct_rtp/   (props changed)
    team/may/ooh323_ipv6_direct_rtp/Makefile
    team/may/ooh323_ipv6_direct_rtp/UPGRADE.txt
    team/may/ooh323_ipv6_direct_rtp/apps/app_celgenuserevent.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_dial.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_directory.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_followme.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_queue.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_skel.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_stack.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_system.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_voicemail.c
    team/may/ooh323_ipv6_direct_rtp/apps/app_voicemail.exports.in
    team/may/ooh323_ipv6_direct_rtp/apps/confbridge/conf_config_parser.c
    team/may/ooh323_ipv6_direct_rtp/build_tools/make_version
    team/may/ooh323_ipv6_direct_rtp/channels/chan_dahdi.c
    team/may/ooh323_ipv6_direct_rtp/channels/chan_iax2.c
    team/may/ooh323_ipv6_direct_rtp/channels/chan_misdn.c
    team/may/ooh323_ipv6_direct_rtp/channels/chan_sip.c
    team/may/ooh323_ipv6_direct_rtp/channels/chan_skinny.c
    team/may/ooh323_ipv6_direct_rtp/channels/chan_unistim.c
    team/may/ooh323_ipv6_direct_rtp/channels/console_board.c
    team/may/ooh323_ipv6_direct_rtp/channels/console_gui.c
    team/may/ooh323_ipv6_direct_rtp/channels/console_video.c
    team/may/ooh323_ipv6_direct_rtp/channels/iax2-parser.c
    team/may/ooh323_ipv6_direct_rtp/channels/iax2-provision.c
    team/may/ooh323_ipv6_direct_rtp/channels/misdn/ie.c
    team/may/ooh323_ipv6_direct_rtp/channels/misdn/isdn_lib.c
    team/may/ooh323_ipv6_direct_rtp/channels/misdn/isdn_msg_parser.c
    team/may/ooh323_ipv6_direct_rtp/channels/misdn/portinfo.c
    team/may/ooh323_ipv6_direct_rtp/channels/misdn_config.c
    team/may/ooh323_ipv6_direct_rtp/channels/sig_analog.c
    team/may/ooh323_ipv6_direct_rtp/channels/sig_pri.c
    team/may/ooh323_ipv6_direct_rtp/channels/sig_ss7.c
    team/may/ooh323_ipv6_direct_rtp/channels/sip/config_parser.c
    team/may/ooh323_ipv6_direct_rtp/channels/sip/dialplan_functions.c
    team/may/ooh323_ipv6_direct_rtp/channels/sip/include/sip.h
    team/may/ooh323_ipv6_direct_rtp/channels/sip/reqresp_parser.c
    team/may/ooh323_ipv6_direct_rtp/channels/sip/sdp_crypto.c
    team/may/ooh323_ipv6_direct_rtp/channels/sip/security_events.c
    team/may/ooh323_ipv6_direct_rtp/channels/sip/srtp.c
    team/may/ooh323_ipv6_direct_rtp/channels/sip/utils.c
    team/may/ooh323_ipv6_direct_rtp/channels/vcodecs.c
    team/may/ooh323_ipv6_direct_rtp/channels/vgrabbers.c
    team/may/ooh323_ipv6_direct_rtp/codecs/codec_dahdi.c
    team/may/ooh323_ipv6_direct_rtp/formats/format_ogg_vorbis.c
    team/may/ooh323_ipv6_direct_rtp/funcs/func_presencestate.c
    team/may/ooh323_ipv6_direct_rtp/funcs/func_realtime.c
    team/may/ooh323_ipv6_direct_rtp/funcs/func_volume.c
    team/may/ooh323_ipv6_direct_rtp/include/asterisk/app.h
    team/may/ooh323_ipv6_direct_rtp/include/asterisk/channel.h
    team/may/ooh323_ipv6_direct_rtp/include/asterisk/config_options.h
    team/may/ooh323_ipv6_direct_rtp/include/asterisk/frame.h
    team/may/ooh323_ipv6_direct_rtp/include/asterisk/strings.h
    team/may/ooh323_ipv6_direct_rtp/main/Makefile
    team/may/ooh323_ipv6_direct_rtp/main/abstract_jb.c
    team/may/ooh323_ipv6_direct_rtp/main/acl.c
    team/may/ooh323_ipv6_direct_rtp/main/alaw.c
    team/may/ooh323_ipv6_direct_rtp/main/aoc.c
    team/may/ooh323_ipv6_direct_rtp/main/app.c
    team/may/ooh323_ipv6_direct_rtp/main/asterisk.c
    team/may/ooh323_ipv6_direct_rtp/main/astfd.c
    team/may/ooh323_ipv6_direct_rtp/main/astmm.c
    team/may/ooh323_ipv6_direct_rtp/main/astobj2.c
    team/may/ooh323_ipv6_direct_rtp/main/audiohook.c
    team/may/ooh323_ipv6_direct_rtp/main/autochan.c
    team/may/ooh323_ipv6_direct_rtp/main/autoservice.c
    team/may/ooh323_ipv6_direct_rtp/main/bridging.c
    team/may/ooh323_ipv6_direct_rtp/main/callerid.c
    team/may/ooh323_ipv6_direct_rtp/main/ccss.c
    team/may/ooh323_ipv6_direct_rtp/main/cdr.c
    team/may/ooh323_ipv6_direct_rtp/main/cel.c
    team/may/ooh323_ipv6_direct_rtp/main/channel.c
    team/may/ooh323_ipv6_direct_rtp/main/channel_internal_api.c
    team/may/ooh323_ipv6_direct_rtp/main/chanvars.c
    team/may/ooh323_ipv6_direct_rtp/main/cli.c
    team/may/ooh323_ipv6_direct_rtp/main/config.c
    team/may/ooh323_ipv6_direct_rtp/main/config_options.c
    team/may/ooh323_ipv6_direct_rtp/main/data.c
    team/may/ooh323_ipv6_direct_rtp/main/datastore.c
    team/may/ooh323_ipv6_direct_rtp/main/db.c
    team/may/ooh323_ipv6_direct_rtp/main/devicestate.c
    team/may/ooh323_ipv6_direct_rtp/main/dial.c
    team/may/ooh323_ipv6_direct_rtp/main/dns.c
    team/may/ooh323_ipv6_direct_rtp/main/dnsmgr.c
    team/may/ooh323_ipv6_direct_rtp/main/dsp.c
    team/may/ooh323_ipv6_direct_rtp/main/enum.c
    team/may/ooh323_ipv6_direct_rtp/main/event.c
    team/may/ooh323_ipv6_direct_rtp/main/file.c
    team/may/ooh323_ipv6_direct_rtp/main/fixedjitterbuf.c
    team/may/ooh323_ipv6_direct_rtp/main/format.c
    team/may/ooh323_ipv6_direct_rtp/main/format_cap.c
    team/may/ooh323_ipv6_direct_rtp/main/format_pref.c
    team/may/ooh323_ipv6_direct_rtp/main/frame.c
    team/may/ooh323_ipv6_direct_rtp/main/framehook.c
    team/may/ooh323_ipv6_direct_rtp/main/fskmodem.c
    team/may/ooh323_ipv6_direct_rtp/main/fskmodem_float.c
    team/may/ooh323_ipv6_direct_rtp/main/fskmodem_int.c
    team/may/ooh323_ipv6_direct_rtp/main/global_datastores.c
    team/may/ooh323_ipv6_direct_rtp/main/hashtab.c
    team/may/ooh323_ipv6_direct_rtp/main/heap.c
    team/may/ooh323_ipv6_direct_rtp/main/image.c
    team/may/ooh323_ipv6_direct_rtp/main/indications.c
    team/may/ooh323_ipv6_direct_rtp/main/io.c
    team/may/ooh323_ipv6_direct_rtp/main/jitterbuf.c
    team/may/ooh323_ipv6_direct_rtp/main/libasteriskssl.c
    team/may/ooh323_ipv6_direct_rtp/main/loader.c
    team/may/ooh323_ipv6_direct_rtp/main/lock.c
    team/may/ooh323_ipv6_direct_rtp/main/logger.c
    team/may/ooh323_ipv6_direct_rtp/main/manager.c
    team/may/ooh323_ipv6_direct_rtp/main/message.c
    team/may/ooh323_ipv6_direct_rtp/main/netsock.c
    team/may/ooh323_ipv6_direct_rtp/main/netsock2.c
    team/may/ooh323_ipv6_direct_rtp/main/pbx.c
    team/may/ooh323_ipv6_direct_rtp/main/plc.c
    team/may/ooh323_ipv6_direct_rtp/main/presencestate.c
    team/may/ooh323_ipv6_direct_rtp/main/privacy.c
    team/may/ooh323_ipv6_direct_rtp/main/rtp_engine.c
    team/may/ooh323_ipv6_direct_rtp/main/say.c
    team/may/ooh323_ipv6_direct_rtp/main/sched.c
    team/may/ooh323_ipv6_direct_rtp/main/security_events.c
    team/may/ooh323_ipv6_direct_rtp/main/slinfactory.c
    team/may/ooh323_ipv6_direct_rtp/main/srv.c
    team/may/ooh323_ipv6_direct_rtp/main/stdtime/localtime.c
    team/may/ooh323_ipv6_direct_rtp/main/strcompat.c
    team/may/ooh323_ipv6_direct_rtp/main/strings.c
    team/may/ooh323_ipv6_direct_rtp/main/stun.c
    team/may/ooh323_ipv6_direct_rtp/main/syslog.c
    team/may/ooh323_ipv6_direct_rtp/main/taskprocessor.c
    team/may/ooh323_ipv6_direct_rtp/main/tcptls.c
    team/may/ooh323_ipv6_direct_rtp/main/tdd.c
    team/may/ooh323_ipv6_direct_rtp/main/term.c
    team/may/ooh323_ipv6_direct_rtp/main/test.c
    team/may/ooh323_ipv6_direct_rtp/main/threadstorage.c
    team/may/ooh323_ipv6_direct_rtp/main/timing.c
    team/may/ooh323_ipv6_direct_rtp/main/translate.c
    team/may/ooh323_ipv6_direct_rtp/main/udptl.c
    team/may/ooh323_ipv6_direct_rtp/main/ulaw.c
    team/may/ooh323_ipv6_direct_rtp/main/utils.c
    team/may/ooh323_ipv6_direct_rtp/main/xml.c
    team/may/ooh323_ipv6_direct_rtp/main/xmldoc.c
    team/may/ooh323_ipv6_direct_rtp/pbx/dundi-parser.c
    team/may/ooh323_ipv6_direct_rtp/pbx/pbx_config.c
    team/may/ooh323_ipv6_direct_rtp/res/ael/pval.c
    team/may/ooh323_ipv6_direct_rtp/res/res_adsi.c
    team/may/ooh323_ipv6_direct_rtp/res/res_config_odbc.c
    team/may/ooh323_ipv6_direct_rtp/res/res_fax.c
    team/may/ooh323_ipv6_direct_rtp/res/res_odbc.c
    team/may/ooh323_ipv6_direct_rtp/res/res_smdi.c
    team/may/ooh323_ipv6_direct_rtp/res/res_speech.c
    team/may/ooh323_ipv6_direct_rtp/res/snmp/agent.c
    team/may/ooh323_ipv6_direct_rtp/tests/test_config.c
    team/may/ooh323_ipv6_direct_rtp/tests/test_voicemail_api.c
    team/may/ooh323_ipv6_direct_rtp/utils/astdb2bdb.c
    team/may/ooh323_ipv6_direct_rtp/utils/astdb2sqlite3.c

Propchange: team/may/ooh323_ipv6_direct_rtp/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.

Propchange: team/may/ooh323_ipv6_direct_rtp/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Mon Jul  2 16:56:44 2012
@@ -1,2 +1,2 @@
 /team/mmichelson/private/phones-trunk:358764-361321
-/trunk:331201-331202,346391,354429,356042,357272,360190,362888,362919-362920,368421-368569
+/trunk:331201-331202,346391,354429,356042,357272,360190,362888,362919-362920,368421-368569,368588-369034

Propchange: team/may/ooh323_ipv6_direct_rtp/
            ('svnmerge-blocked' removed)

Propchange: team/may/ooh323_ipv6_direct_rtp/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Jul  2 16:56:44 2012
@@ -1,1 +1,1 @@
-/trunk:1-313481,313483-313906,313908-313943,313945-368584
+/trunk:1-368158,368588-369034

Modified: team/may/ooh323_ipv6_direct_rtp/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_ipv6_direct_rtp/Makefile?view=diff&rev=369540&r1=369539&r2=369540
==============================================================================
--- team/may/ooh323_ipv6_direct_rtp/Makefile (original)
+++ team/may/ooh323_ipv6_direct_rtp/Makefile Mon Jul  2 16:56:44 2012
@@ -497,40 +497,23 @@
 
 NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
 OLDHEADERS=$(filter-out $(NEWHEADERS) $(notdir $(DESTDIR)$(ASTHEADERDIR)),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
+INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTETCDIR)" "$(ASTVARRUNDIR)" \
+	"$(ASTSPOOLDIR)" "$(ASTSPOOLDIR)/dictate" "$(ASTSPOOLDIR)/meetme" \
+	"$(ASTSPOOLDIR)/monitor" "$(ASTSPOOLDIR)/system" "$(ASTSPOOLDIR)/tmp" \
+	"$(ASTSPOOLDIR)/voicemail" "$(ASTHEADERDIR)" "$(ASTHEADERDIR)/doxygen" \
+	"$(ASTLOGDIR)" "$(ASTLOGDIR)/cdr-csv" "$(ASTLOGDIR)/cdr-custom" \
+	"$(ASTLOGDIR)/cel-custom" "$(ASTDATADIR)" "$(ASTDATADIR)/documentation" \
+	"$(ASTDATADIR)/documentation/thirdparty" "$(ASTDATADIR)/firmware" \
+	"$(ASTDATADIR)/firmware/iax" "$(ASTDATADIR)/images" "$(ASTDATADIR)/keys" \
+	"$(ASTDATADIR)/phoneprov" "$(ASTDATADIR)/static-http" "$(ASTDATADIR)/sounds" \
+	"$(ASTDATADIR)/moh" "$(ASTMANDIR)/man8" "$(AGI_DIR)" "$(ASTDBDIR)"
 
 installdirs:
-	$(INSTALL) -d "$(DESTDIR)$(ASTLIBDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTMODDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSBINDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTETCDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTVARRUNDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)/dictate"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)/meetme"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)/monitor"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)/system"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)/tmp"
-	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)/voicemail"
-	$(INSTALL) -d "$(DESTDIR)$(ASTHEADERDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTHEADERDIR)/doxygen"
-	$(INSTALL) -d "$(DESTDIR)$(ASTLOGDIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTLOGDIR)/cdr-csv"
-	$(INSTALL) -d "$(DESTDIR)$(ASTLOGDIR)/cdr-custom"
-	$(INSTALL) -d "$(DESTDIR)$(ASTLOGDIR)/cel-custom"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/documentation"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/documentation/thirdparty"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/firmware"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/firmware/iax"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/images"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/keys"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/phoneprov"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/static-http"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/sounds"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDATADIR)/moh"
-	$(INSTALL) -d "$(DESTDIR)$(ASTMANDIR)/man8"
-	$(INSTALL) -d "$(DESTDIR)$(AGI_DIR)"
-	$(INSTALL) -d "$(DESTDIR)$(ASTDBDIR)"
+	@for i in $(INSTALLDIRS); do \
+		if [ ! -z "$${i}" -a ! -d "$(DESTDIR)$${i}" ]; then \
+			$(INSTALL) -d "$(DESTDIR)$${i}"; \
+		fi; \
+	done
 
 main-bininstall:
 	+ at DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" ASTLIBDIR="$(ASTLIBDIR)" $(SUBMAKE) -C main bininstall

Modified: team/may/ooh323_ipv6_direct_rtp/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_ipv6_direct_rtp/UPGRADE.txt?view=diff&rev=369540&r1=369539&r2=369540
==============================================================================
--- team/may/ooh323_ipv6_direct_rtp/UPGRADE.txt (original)
+++ team/may/ooh323_ipv6_direct_rtp/UPGRADE.txt Mon Jul  2 16:56:44 2012
@@ -101,6 +101,10 @@
    should be vastly improved.  The HANGUPCAUSE hash should now be used instead
    of SIP_CAUSE. Because of this, the storesipcause option in sip.conf is also
    deprecated.
+ - The sip paramater for Originating Line Information (oli, isup-oli, and
+   ss7-oli) is now parsed out of the From header and copied into the channel's
+   ANI2 information field.  This is readable from the CALLERID(ani2) dialplan
+   function.
 
 chan_unistim
  - Due to massive update in chan_unistim phone keys functions and on-screen 

Modified: team/may/ooh323_ipv6_direct_rtp/apps/app_celgenuserevent.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_ipv6_direct_rtp/apps/app_celgenuserevent.c?view=diff&rev=369540&r1=369539&r2=369540
==============================================================================
--- team/may/ooh323_ipv6_direct_rtp/apps/app_celgenuserevent.c (original)

[... 6897 lines stripped ...]



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