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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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  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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Make hint watcher callback take const strings for context and exten parameters.
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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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delete unproperly changed svn props
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.
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  r328446 | lmadsen | 2011-07-15 16:41:12 -0400 (Fri, 15 Jul 2011) | 1 line
  
  Revert changes to defaultenabled state for modules in Asterisk 1.8
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Update UPGRADE.txt and CHANGES files.
Update documentation files stating that deprecated modules are no longer built by default.
Build app_macro by default because things depend on it.
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  r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines
  
  Typo
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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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  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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  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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  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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  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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  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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Fix mistaken version number