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  r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20 Oct 2011) | 8 lines
  
  add documentation for check_state_unknown in configs/queues.conf.sample
  
  app_queue allows calls to members in a "Unknown" state to be treated as available
  setting check_state_unknown = yes will cause app_queue to query the channel driver
  to better determine the state this only applies to queues with ringinuse or ignorebusy
  set appropriately. 
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  r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines
  
  Add option to check state when state is unknown
  
  r341486 reverts r325483 this is a rework of the patch.
  optimize to minimize load.
  
  add option check_state_unknown to control whether a member with unknown
  device state is checked there is a small % chance that calls will be sent
  to the member when they on a call.
  
  app_queue will see a device with unknown state as available and does not 
  try verify the state without this option enabled.
  
  Review: https://reviewboard.asterisk.org/r/1535/
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The code was originally copied from the is_int() function in the AEL
code. wdoekes pointed out that the function should take a const char*
and that their was an unneeded variable. This is now fixed.
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  r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct 2011) | 18 lines
  
  Fix a performance regression introduced in r325483.
  
  The regression was caused by a call to ast_parse_device_state() in app_queue's
  ring_entry() function. The ast_parse_device_state() function eventually calls
  ast_channel_get_full() with a channel name prefix which causes it to walk the
  channel list causing massive lock contention and slow downs.
  
  This patch fixes the regression by removing the call to
  ast_parase_device_state() which should be unnecessary. Queue member device
  state should be maintained by device state events. Some users have seen
  instances where busy agents were called when they shouldn't have, which is the
  reason the call to ast_parse_device_state() was added. That change appears to
  have resolved that issue but also causes this performance regression. There may
  still be issues with queue member status, and if so, alternative methods should
  be investigated to resolve them.
  
  AST-695
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Google has recently make some changes (again) to their protocol.  Rather then
patching asterisk to flip between the two different methods, we now allow both.

Lets hope this keeps Google Voice happy for a while.

(closes issue ASTERISK-18714)
Reported by: Iordan Iordanov
Patches:
    chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311)
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Just create an normal API function in strings.h that does the same thing
just to be safe.

ASTERISK-17146
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Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS
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If a SIP dial string contains a numeric hostname that is not a peer name,
don't try to resolve it as it is unlikely that someone really means
Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that
create_addr returns -1 if an address isn't resolved so that we don't
attempt to send SIP requests to an address that doesn't resolve.

(closes issue ASTERISK-17146, ASTERISK-17716)

Review: https://reviewboard.asterisk.org/r/1532/
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* Fix potential deadlocks in SIP and IAX blind transfer to parking.

* Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter).  Created ast_park_call_exten() and ast_masq_park_call_exten()
to maintian API compatibility.

* Made masq_park_call() handle a failed ast_channel_masquerade() setup.

* Reduced excessive struct parkeduser.peername[] size.
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Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to
test the commit message.

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If parse_uri was called with an empty URI, some pointers would be
modified and an invalid read could result. This patch avoids calling
parse_uri with an empty contact uri when parsing REGISTER requests. 

AST-2011-012

(closes issue ASTERISK-18668)
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........
  r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines
  
  Voicemail compiler flags are 'core' support
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  r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines
  
  Fix previous commit
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(issue ASTERISK-18680)
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(closes issue ASTERISK-18696)
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whether modules are embedded or not; using just the bare category name led to
accidentally enabling these options when users used the wrong "--enable"
operation on the menuselect command line.

Now the internal option names are prefixed with "EMBED_", so they won't be
the same as the name of the category containing the modules they control
the embedding of.
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If a simple switch was started on a device and then a specific call
made (such as redial or speed dial), on timeout of the simple switch
the call would be attempted again. This patch only allows the simple
switch to make a call if the substate is still in the collecting
digits mode.

Also added small debug message to dialAndAactivate sub. 

Tested by snuff and myself.



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  r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
  
  Merged revisions 340970 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
    
    Quiet RTCP Receiver Reports during fax transmission
    
    RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
    The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
    code was added to support the bug fix.
    
    (closes issue ASTERISK-18400)
  ........
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(issue ASTERISK-18268)
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Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
displaying a WARNING message.

(closes issue ASTERISK-18610)
 Patch by: Kristijan_Vrban
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Party A calls Party B.
Party A DTMF blind transfers Party B to Party C.
Party A channel continues to execute dialplan.

* Fixed the return value of builtin_blindtransfer() to return the correct
value after a transfer so the dialplan will not keep executing.

* Removed unnecessary connected line update that did not really do
anything.

* Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer().

* Fixed leak of xferchan for failure cases in check_goto_on_transfer().

* Updated debug messages in builtin_blindtransfer() and
check_goto_on_transfer().

(closes issue ASTERISK-18275)
Reported by: rmudgett
Tested by: rmudgett
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  r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines
  
  Only change the capabilities on the gateway when
  the session is been destroyed there is still
  a race condition that ends in a segfault.
  
  if the caps are changed the logic in res_fax_spandsp
  will run T30 code not gateway code to end the session.
  this has been experienced on a "slower" under spec system.
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  r340718 | schmidts | 2011-10-13 06:59:50 +0000 (Thu, 13 Oct 2011) | 9 lines
  
  Merged revisions 340717 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines
    
    storing the route-set also on a 181 response not only on 180,182 or 183.
  ........
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Avoid possible jump based on unitialized value
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There is no documented reason to not add the query field to the varlist
returned by a realtime multi query, despite the config category being
set to its value. Of course, there is no documentation that the category
should be set to the value either. There is lots of no documentation
when it comes to realtime. But, other engines do not skip this field so
I am forcing this backend to follow the convention, because not doing so
is very silly.
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  r340577 | schmidts | 2011-10-12 20:33:37 +0000 (Mit, 12 Okt 2011) | 9 lines
  
  Merged revisions 340576 via svnmerge from 
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  ........
    r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines
    
    Store route-set from provisional SIP responses so early-dialog requests can be routed properly
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  r340578 | twilson | 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) | 16 lines
  
  Merged revisions 340534 via svnmerge from 
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  ........
    r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines
    
    Update SIP realtime fullcontact regardless of caching
    
    We should update the fullcontact field in the realtime table whether or
    not rtcachefriends is set. There is no reason to treat a non-cached
    realtime entity differently than a cached in this regard.
    
    (closes issue ASTERISK-18446)
     Reported by: wdoekes
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The PRI channel alarms were initialized with an inverted sense.

(closes issue ASTERISK-18710)
Reported by: Tzafrir Cohen
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ASTERISK-12175 changed the p and X options to not interfere with the s
option when they are used together.  It makes more sense for the s option
to have priority for the DTMF '*' key since it cannot change its
activation code.  Otherwise, you could not use option s with the p or X
options.

JIRA AST-671
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(closes issue ASTERISK-18612)
Reported by: Tim Osman
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* Added a CLI "ss7 show channels" command that might prove useful for
future debugging.

* Made the incoming SS7 channel event check and gripe message uniform.

* Made sure that the DNID string for an incoming call is always
initialized.

(issue ASTERISK-17966)
Reported by: Kenneth Van Velthoven
Patches:
      jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett
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* Fixed deadlock potential calling dialog_unlink_all() in
__sip_autodestruct().  Found by helgrind.

* Fixed deadlock potential in handle_request_invite() after calling
sip_new().  Found by helgrind.

* The sip_new() function now returns with the created channel already
locked.

* Removed the dead code that starts a PBX in in sip_new().  No sip_new()
callers caused that code to be executed and it was a bad thing to do
anyway.

* Removed unused parameters and return value from dialog_unlink_all().

* Made dialog_unlink_all() and __sip_autodestruct() safely obtain the
owner and private channel locks without a deadlock avoidance loop.
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RFC 6234 is an update to RFC 3174 from which the code was originally taken.
It has a slightly better code, and a better phrased license (simple 3-clause
BSD).

* main/sha1.c is sha1.c from RFC 6234 with formatting changes only.
* include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234.
* Removed unused include of asterisk/sha1.h from main/channels.c

Review: https://reviewboard.asterisk.org/r/1503/

Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263

Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280


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* Fixed race between calling an AMI action callback and unregistering that
action.  Refixes ASTERISK-13784 broken by ASTERISK-17785 change.

* Fixed potential memory leak if an AMI action failed to get registered
because is already was registered.  Part of the ao2 conversion.

* Fixed AMI ListCommands action not walking the actions list with a lock
held.

* Fix usage of ast_strdupa() and alloca() in loops.  Excess stack usage.

* Fix AMI Originate action Variable header requiring a space after the
header colon.  Reported by Yaroslav Panych on the asterisk-dev list.

* Increased the number of listed variables allowed per AMI Originate
action Variable header to 64.

* Fixed AMI GetConfigJSON action output format.

* Fixed usage of res contents outside of scope in append_channel_vars().

* Fixed inconsistency of config file channelvars option.  The values no
longer accumulate with every channelvars option in the config file.  Only
the last value is kept to be consistent with the CLI "manager show
settings" command.

(closes issue ASTERISK-18479)
Reported by: Jaco Kroon
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(closes issue AST-654)


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  r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011) | 8 lines
  
  On astdb conversion, also warn about permissions requirements
  
  The user running Asterisk must have permission to the directory
  the Asterisk database resides in since SQLite 3 needs to be able
  to create a journal file.
  
  (closes issue ASTERISK-18174)
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  r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) | 8 lines
  
  Add astdb conversion utility for Berkeley to SQLite 3
  
  If someone wants to backtrack from Asterisk 1.8 to 10 they can use the
  astdb2bdb utility to convert the database back to the Berkeley format
  that Asterisk 1.8 uses.
  
  Review: https://reviewboard.asterisk.org/r/1502/
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  r340220 | twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
  
  Add a missing file for the astdb2bdb conversion utility
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