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  r336734 | tilghman | 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) | 18 lines
  
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  ........
    r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines
    
    Various changes to allow 1.8 to compile on Mac OS X Lion (10.7)
    
    * Makefile workaround for 10.6 extended to work on 10.7 and later.
    * Now uses the 'weak' symbol for Lion systems, which no longer support
      'weak_import'
    
    Closes ASTERISK-17612.
    Closes ASTERISK-18213.
    
    Tested by: tilghman, oej.
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  r336717 | jrose | 2011-09-19 15:16:23 -0500 (Mon, 19 Sep 2011) | 14 lines
  
  Merged revisions 336716 via svnmerge from 
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    r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines
    
    Document applications that play audio and do not answer unanswered calls.
    
    This patch is part of an effort to document early media and its usage. If you are
    interested in contributing to this documentation effort, there are probably other
    applications worth documenting as well as an Asterisk wiki article at
    https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
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  r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines
  
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  ........
    r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines
    
    Made Dial d and H options no longer immediately auto-answer the calling leg.
    
    The Dial d and H options break DTMF attended transfer atxferdropcall
    option.
    
    1) Party A calls party B.
    2) Party B does a DTMF attended transfer to Party C.
    
    If the dialplan uses the Dial d or H options to call Party C then the Dial
    application answers the call immediately before initiating the call leg to
    Party C.  The premature answer causes the transfer code to not invoke the
    atxferdropcall=no behavior for a blonde transfer since Party C has
    "answered".  The transfer code thinks that Party B has "consulted" with
    Party C when Party B hangs up and completes the transfer to Party A.
    Party A now hears ringback until Party C actually answers.
    
    ASTERISK-13294 Dial d option.
    ASTERISK-11067 Dial H option to disconnect before answer.
    
    The referenced issues made Dial answer with the d and H options because
    many SIP and ISDN phones cannot send DTMF before the call is connected.
    
    * Made require the dialplan to control when or if the call needs to be
    answered to use the Dial application d and H options.  (The call is no
    longer surprise answered when using the Dial d or H options.)
    
    Review: https://reviewboard.asterisk.org/r/1381/
    
    JIRA AST-623
    JIRA AST-666
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  r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7 lines
  
  Update get_ilbc_source.sh script to work again.
  
  Recently iLBC support in Asterisk has changed after the acquisition of GIPS
  by Google. More information about how this may affect you is available in a
  blog post at:
  
    http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
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  r336570 | rmudgett | 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) | 11 lines
  
  Merged revisions 336569 via svnmerge from 
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  ........
    r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines
    
    Rework sig_pri_hangup() to be simpler and clearer.
    
    JIRA AST-675
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  r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån, 19 Sep 2011) | 12 lines
  
  Merged revisions 336501 via svnmerge from 
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    r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines
    
    Add diversion header to a 302 redirect response if we have diversion data 
    
    (closes issue ASTERISK-18143)
    	patch by oej
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  r336500 | irroot | 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) | 19 lines
  
  Merged revisions 336499 via svnmerge from 
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  ........
    r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines
    
    A long time ago in a galaxy far far away a IPv6 update was made,
    chan_h323 was not updated causeing all to flee to chan_ooh323.
    
    the brave Jedi [asterisk developers] pondered this miscarrige of justice
    and restored order to the force for the sake of closing out 2 old issues.
    
    (closes issue ASTERISK-17278)
    (closes issue ASTERISK-17500)
    Reported by: dread, sybasesql
    Tested by: irroot
    Reviewed by: IRC (russellb, kpfleming)
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  r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån, 19 Sep 2011) | 9 lines
  
  Merged revisions 336440 via svnmerge from 
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    r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 lines
    
    Make sure manager_debug option is reset at reload
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  r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån, 19 Sep 2011) | 16 lines
  
  Merged revisions 336378 via svnmerge from 
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    r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines
    
    Add missing unlock at MWI message sending time
    
    (closes issue ASTERISK-18573)
    
    Patches:
       sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky
    
    Thanks to irrot for the reminder, to Gregory for the patch!
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  r336316 | twilson | 2011-09-16 17:11:39 -0500 (Fri, 16 Sep 2011) | 9 lines
  
  Merged revisions 336314 via svnmerge from 
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    r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines
    
    Whitespace fix
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  r336313 | twilson | 2011-09-16 17:07:00 -0500 (Fri, 16 Sep 2011) | 12 lines
  
  Merged revisions 336312 via svnmerge from 
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  ........
    r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines
    
    Add missing frame types to func_frame_trace
    
    Also casts control frames to the proper enum so that the compile will catch
    new additions.
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  r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines
  
  Merged revisions 336294 via svnmerge from 
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  ........
    r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines
    
    Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
    
    In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
    break when starting a call with directmedia. This patch queues a new type of control frame
    so that our RTP bridge loop can properly detect when these situations occur and check to see
    if peers need to be updated in order to send their media to the proper location.
    
    (Closes issue ASTERISK-18340)
    Reported by: Thomas Arimont
    (Closes issue ASTERISK-17725)
    Reported by: kwk
    Tested by: twilson, jrose
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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 
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  ........
    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 
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  ........
    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 
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  ........
    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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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>
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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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  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
  
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    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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335385 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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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
  
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    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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335349 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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
  
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    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 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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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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