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  r336235 | seanbright | 2011-09-16 15:10:39 -0400 (Fri, 16 Sep 2011) | 9 lines
  
  Merged revisions 336234 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines
    
    Make a note that inotify won't work with an NFS mounted spooler directory.
  ........
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  r336167 | irroot | 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) | 22 lines
  
  Merged revisions 336166 via svnmerge from 
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  ........
    r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines
    
    The round robin routing routine in chan_misdn.c is broken.
    
    it rotates between ports but never checks the channels in the ports.
    
    i have extensivly tested it and verified it works on 1 upto 4 ports.
    before the patch only 1 out of each port was used now all are used as
    expected.
    
    (closes issue ASTERISK-18413)
    Reported by: irroot
    Tested by: irroot
    Reviewed by: irroot
        
    Review: https://reviewboard.asterisk.org/r/1410/
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  r336094 | irroot | 2011-09-15 17:54:46 +0200 (Thu, 15 Sep 2011) | 26 lines
  
  Merged revisions 336093 via svnmerge from 
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  ........
    r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines
    
    
    Locking order in app_queue.c causes deadlocks.
    
    a channel lock must never be held with the queues container lock held.
    
    the deadlock occured on masquerade.
    
    the queues container lock is a relic of the past the old queue module lock.
    with ao2 there is no need to hold this lock when dealing with members this
    patch removes unneeded locks.
    
    (closes issue ASTERISK-18101)
    (closes issue ASTERISK-18487)
    Reported by: Paul Rolfe, Jason Legault
    Tested by: irroot, Jason Legault, Paul Rolfe
    Reviewed by: Matthew Nicholson
    
    Review: https://reviewboard.asterisk.org/r/1402/
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  r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011) | 2 lines
  
  Removes some no-op code found in format_cap.c.
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  r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
  
  Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
  
  When using Meetme as a modular call bridge from third party applications, it's handy to make
  it behave like a normal call bridge. When the second to last person exists, the last person
  will be kicked out of the conference when this option is enabled.
  
  (closes issue ASTERISK-18234)
  
  Review: https://reviewboard.asterisk.org/r/1376/
  
  Patch by oej, sponsored by ClearIT, Solna, Sweden
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  r335991 | irroot | 2011-09-15 10:29:12 +0200 (Thu, 15 Sep 2011) | 17 lines
  
  Merged revisions 335978 via svnmerge from 
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  ........
    r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines
    
    lock the channel before calling ast_bridged_channel() to prevent a seg fault.
    
    AMI agents list called on shutdown causes a segfault, introducing proper locking
    will prevent this.
    
    (closes issue ASTERISK-18092)
    
    Reported by: agustina
    Patches: chan_agent.patch (License #5041) patch uploaded by irroot
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  r335912 | rmudgett | 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) | 20 lines
  
  Merged revisions 335911 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines
    
    Remove unnecessary libpri dependency checks in the configure script.
    
    Using the --with-pri option with the configure script generated an error
    about not having PRI_L2_PERSISTENCE if you did not have the absolute
    latest libpri SVN checkout installed.
    
    The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to
    be for libraries that are dependent upon other libraries and not
    necessarily for optional/added features within a library.
    
    (closes issue ASTERISK-18535)
    Reported by: Michael Keuter
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  r335852 | rmudgett | 2011-09-14 11:00:37 -0500 (Wed, 14 Sep 2011) | 18 lines
  
  Merged revisions 335851 via svnmerge from 
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  ........
    r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines
    
    Fixed cut-n-paste regression using the wrong variable.
    
    Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
    sections for channel configuration.
    
    (closes issue ASTERISK-18496)
    Reported by: Sean Darcy
    Patches:
          jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: Sean Darcy, rmudgett
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  r335791 | mnicholson | 2011-09-14 08:28:50 -0500 (Wed, 14 Sep 2011) | 11 lines
  
  Merged revisions 335790 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines
    
    The tech and data members of fast_originate_helper are not string fields.
    
    ASTERISK-17709
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  r335721 | rmudgett | 2011-09-13 17:10:44 -0500 (Tue, 13 Sep 2011) | 9 lines
  
  Merged revisions 335720 via svnmerge from 
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  ........
    r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line
    
    Remove obsolete todo comment about PICKUPRESULT.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335722 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1432/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335719 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Do parse the option "defaultlanguage" from the [options] section of
asterisk.conf, as in the sample config file. Otherwise the build-time
default language (normally "en") is always the default one.

Review: https://reviewboard.asterisk.org/r/1342/


Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>
Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716
Original-Commit: http://svn.digium.com/svn/asterisk/branches/10@335717

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  r335656 | tilghman | 2011-09-13 13:55:33 -0500 (Tue, 13 Sep 2011) | 11 lines
  
  Merged revisions 335655 via svnmerge from 
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  ........
    r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines
    
    Move mandatory checks closer to the beginning of the file.
    
    If these are going to fail, they should fail as quickly as possible.
  ........
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  r335653 | mnicholson | 2011-09-13 13:47:57 -0500 (Tue, 13 Sep 2011) | 12 lines
  
  Merged revisions 335618 via svnmerge from 
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  ........
    r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines
    
    Don't limit the size of appdata for manager originate actions.
    
    ASTERISK-17709
    Patch by: tilghman (with modifications)
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Review: https://reviewboard.asterisk.org/r/1434/


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Review: https://reviewboard.asterisk.org/r/1427/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1432/


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  r335510 | russell | 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) | 22 lines
  
  Merged revisions 335497 via svnmerge from 
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  ........
    r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines
    
    Fix a crash in res_ais.
    
    This patch resolves a crash observed in a load testing environment that
    involved the use of the res_ais module.  I observed some crashes where
    the event delivery callback would get called, but the length parameter
    incidcating how much data there was to read was 0.  The code assumed
    (with good reason I would think) that if this callback got called, there
    was an event available to read.  However, if the rare case that there's
    nothing there, catch it and return instead of blowing up.
    
    More specifically, the change always ensure that the size of the received
    event in the cluster is always big enough to be a real ast_event.
    
    Review: https://reviewboard.asterisk.org/r/1423/
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  r335434 | mnicholson | 2011-09-12 10:55:48 -0500 (Mon, 12 Sep 2011) | 13 lines
  
  Merged revisions 335433 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines
    
    Properly set caller_warning and callee_warning before we try to use them.
    
    ASTERISK-18199
    Patch by: elguero
    Testing by: rtang
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  r335346 | kmoore | 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) | 17 lines
  
  Merged revisions 335341 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines
    
    Ensure frames are not written to dialed channel if ringback is requested
    
    When a single channel was dialed and there was media to be forwarded to the
    calling channel, the media was written without regard for ringback causing
    silence to be heard in some circumstances.  This regression was introduced
    when the meaning of "single" changed to mean only the number of channels
    dialed.
    
    (closes issue ASTERISK-18083)
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Review: https://reviewboard.asterisk.org/r/1429/

(closes issue ASTERISK-18497)

Thanks to russellb for peer review.


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  r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines
  
  Merged revisions 335319 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines
    
    Lock the peer->mvipvt to avoid crashes with SIP history enabled
    
    After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
    which cause issues with SIP history additions in combination with the max limit for
    number of history entries.
    
    Review: https://reviewboard.asterisk.org/r/1373/
    
    (closes issue ASTERISK-18288)
    
    Thanks to irrot for peer review. Work with this bug funded by IPvision AS
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  r335321 | kmoore | 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) | 16 lines
  
  Merged revisions 335320 via svnmerge from 
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  ........
    r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines
    
    Prevent IAX2 from getting IPv6 addresses via DNS
    
    IAX2 does not support IPv6 and getting such addresses from DNS can cause error
    messages on the remote end involving bad IPv4 address casts in the presence of
    IPv6/IPv4 tunnels.  This patch ensures that IAX2 will not encounter IPv6
    addresses via DNS queries.
    
    (closes issue ASTERISK-18090)
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  r335260 | schmidts | 2011-09-12 11:11:45 +0000 (Mon, 12 Sep 2011) | 12 lines
  
  Merged revisions 335259 via svnmerge from 
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  ........
    r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines
    
    build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value.
    adding an ao2_unlink from the peers_by_ip container fix it.
    
    Review: https://reviewboard.asterisk.org/r/1428/
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Review: https://reviewboard.asterisk.org/r/1426/


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  r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
  
  Merged revisions 335064 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
    
    Updated SIP 484 handling; added Incomplete control frame
    
    When a SIP phone uses the dial application and receives a 484 Address 
    Incomplete response, if overlapped dialing is enabled for SIP, then
    the 484 Address Incomplete is forwarded back to the SIP phone and the
    HANGUPCAUSE channel variable is set to 28.  Previously, the Incomplete
    application dialplan logic was automatically triggered; now, explicit
    dialplan usage of the application is required.
    
    Additionally, this patch adds a new AST_CONTOL_FRAME type called
    AST_CONTROL_INCOMPLETE.  If a channel driver receives this control frame,
    it is an indication that the dialplan expects more digits back from the
    device.  If the device supports overlap dialing it should attempt to 
    notify the device that the dialplan is waiting for more digits; otherwise,
    it can handle the frame in a manner appropriate to the channel driver.
    
    (closes issue ASTERISK-17288)
    Reported by: Mikael Carlsson
    Tested by: Matthew Jordan
    
    Review: https://reviewboard.asterisk.org/r/1416/
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  r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) | 9 lines
  
  
  Move code for VALID_EXTEN from app_readexten to func_dialplan
  
  Mark VALID_EXTEN deprecated.
  
  Review: https://reviewboard.asterisk.org/r/1396/
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  r334954 | rmudgett | 2011-09-08 17:28:56 -0500 (Thu, 08 Sep 2011) | 17 lines
  
  Merged revisions 334953 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines
    
    Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used.
    
    Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to
    unregister its logger level.
    
    * Make ast_logger_unregister_level() use ast_free() instead of free().
    When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call
    to free().  Therefore, if you allocated memory with a form of ast_malloc
    you must free it with ast_free.
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  r334844 | pabelanger | 2011-09-07 15:37:24 -0400 (Wed, 07 Sep 2011) | 11 lines
  
  Merged revisions 334843 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines
    
    Cleanup chan_iax2.c log messages
    
    Review: https://code.asterisk.org/code/cru/CR-AST-11
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  r334841 | rmudgett | 2011-09-07 14:33:38 -0500 (Wed, 07 Sep 2011) | 17 lines
  
  Merged revisions 334840 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines
    
    Fix AMI action Park crash.
    
    * Made AMI action Park not say anything to the parker channel (AMI header
    Channel2) since the AMI action is a third party parking the call.  (This
    is a change in behavior that cannot be preserved without a lot of effort.)
    
    * Made not play pbx-parkingfailed if the Park 's' option is used.
    
    JIRA AST-660
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  r334747 | schmidts | 2011-09-07 15:10:37 +0000 (Wed, 07 Sep 2011) | 9 lines
  
  Merged revisions 334682 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines
    
    Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
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  r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines
  
  Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334712 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334683 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/10

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  r334621 | alecdavis | 2011-09-07 20:14:50 +1200 (Wed, 07 Sep 2011) | 9 lines
  
  Merged revisions 334620 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines
    
    peroid typo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334623 65c4cc65-6c06-0410-ace0-fbb531ad65f3