remove 1.8 merge properties

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@328075 f38db490-d61c-443f-a65b-d21fe96a405b
set 1.10 merge properties

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@328076 f38db490-d61c-443f-a65b-d21fe96a405b
Add UPGRADE-1.10.txt file from UPGRADE.txt.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@328079 f38db490-d61c-443f-a65b-d21fe96a405b
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  r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines
  
  Preserve sample rate quality of wideband mixmonitor recordings.
  
  MixMonitor has the ability to record in any file format Asterisk supports,
  but the quality of wideband audio is not preserved.  This is because
  regardless of the sample rate the call is being recorded in, the audio
  is always downsampled to 8khz and then upsampled to whatever wideband
  format it is being written as.  This patch resolves this by requesting
  the audio from the audiohook in the signed linear format closest to the
  sample rate of the format we are writing.  This fix is only possible for
  Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband
  audio.
  
  Review: https://reviewboard.asterisk.org/r/1314/
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  r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines
  
  tune the v21 preamble detector to properly detect the desired sequence of hits
  and misses
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  r328207 | jrose | 2011-07-14 14:45:18 -0500 (Thu, 14 Jul 2011) | 13 lines
  
  Merged revisions 328205 via svnmerge from 
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    r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines
    
    Monitor application arguments requirements fixed.
    
    Monitor was requiring options in spite of no individual option on Monitor being required.
    
    Review: https://reviewboard.asterisk.org/r/1320/
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  r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
  
  Merged revisions 328209 via svnmerge from 
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    r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
    
    Introduce <support_level> tags in MODULEINFO.
    This change introduces MODULEINFO into many modules in Asterisk in order to show
    the community support level for those modules. This is used by changes committed
    to menuselect by Russell Bryant recently (r917 in menuselect). More information about
    the support level types and what they mean is available on the wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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  r328317 | rmudgett | 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) | 13 lines
  
  Merged revisions 328302 via svnmerge from 
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    r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines
    
    Missing SIP pvt and channel unlock in sip_set_rtp_peer().
    
    Regression introduced by -r326144.
    
    Add missing SIP pvt and channel unlock in sip_set_rtp_peer().
  ........
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  r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines
  
  Make hint watcher callback take const strings for context and exten parameters.
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@328344 f38db490-d61c-443f-a65b-d21fe96a405b
Add SLA to skinny.

Adds sublines to skinny lines. Each subline can be attached to an 
SLA station/trunk combo. Includes the following functionality:

Callid is persistent for both in/out calls on all skinny devices.
Can join, hold, resume.
All sublines appear under a single line button.

See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc.

(closes issue ASTERISK-17947)

Review: https://reviewboard.asterisk.org/r/1239/



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  r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) | 2 lines
  
  Update UPGRADE.txt and CHANGES files.
  Update documentation files stating that deprecated modules are no longer built by default.
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  r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) | 1 line
  
  Build app_macro by default because things depend on it.
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  r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, 15 Jul 2011) | 13 lines
  
  Merged revisions 328427 via svnmerge from 
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    r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines
    
    small gk processing fixes:
    - decrease for 1 second registration ttl for very low expirations (some
      providers expire few earlier than TTL)
    - delete rrq and registration expire timers on URQ received as we make
      new registration.
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  r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, 15 Jul 2011) | 2 lines
  
  delete unproperly changed svn props
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@328502 f38db490-d61c-443f-a65b-d21fe96a405b
Merged revisions 328541 via svnmerge from 
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  r328541 | tilghman | 2011-07-18 02:11:26 -0500 (Mon, 18 Jul 2011) | 9 lines
  
  Merged revisions 328540 via svnmerge from 
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    r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines
    
    Typo
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  r328609 | markm | 2011-07-18 08:37:53 -0400 (Mon, 18 Jul 2011) | 15 lines
  
  Merged revisions 328593 via svnmerge from 
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    r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines
    
    Fixed invalid read and null pointer deref on asterisk shutdown.
    
    In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash.
    
    (closes issue ASTERISK-17927)
    Reported by: Mark Murawski
    Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher
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  r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
  
  Merged revisions 328608 via svnmerge from 
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    r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
    
    If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
    
    Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure.  But this will fix a crash.
    
    (closes issue ASTERISK-17909)
    Reported by: Mark Murawski
    Tested by: Mark Murawski
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  r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines
  
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    r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines
    
    app_dial may double free a channel datastore
    
    When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash.  Make sure to check if the datastore still exists before trying to free it.
    
    (closes issue ASTERISK-17917)
    Reported by: Mark Murawski
    Tested by: Mark Murawski
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  r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines
  
  Merged revisions 328716 via svnmerge from 
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    r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines
    
    Make AST_LIST_REMOVE safer
    
    AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This
    commit also adds linked list unit tests.
    
    Review: https://reviewboard.asterisk.org/r/1321/
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  r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines
  
  Merged revisions 328770 via svnmerge from 
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    r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines
    
    MeetMe requests a PIN twice in some circumstances
    
    If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
    options, MeetMe will ask for the PIN two times: once for creating the
    conference and once for entering the conference.  This behavior was introduced
    in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
    controlling PIN entry for joining a conference.
    
    (closes AST-601)
    Review: https://reviewboard.asterisk.org/r/1305/
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  r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
  
  Merged revisions 328823 via svnmerge from 
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    r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
    
    RTP bridge away with inband DTMF and feature detection
    
    When deciding whether Asterisk was allowed to bridge the call away from the
    core, chan_sip did not take into account the usage of features on dialed
    channels that require monitoring of DTMF on channels utilizing inband DTMF.
    This would cause Asterisk to allow the call to be locally or remotely bridged, 
    preventing access to the data required to detect activations of such features.
    
    (closes 17237)
    Review: https://reviewboard.asterisk.org/r/1302/
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