When I moved the ARI WebSocket from /ws to /ari/events, I added code to
allow a WebSocket to connect without specifying the subprotocol if
there's only one subprotocol handler registered for the WebSocket.

Naively, I coded it to always respond with the subprotocol in use.
Unfortunately, according to RFC 6455, if the server's response includes
a subprotocol header field that "indicates the use of a subprotocol that
was not present in the client's handshake [...], the client MUST _Fail
the WebSocket Connection_.", emphasis theirs.

This patch correctly omits the Sec-WebSocket-Protocol if one is not
specified by the client.

(closes issue ASTERISK-22441)
Review: https://reviewboard.asterisk.org/r/2828/
........

Merged revisions 399039 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change ensures that MeetMeAdmin commands requiring a user actually
get a user and fixes another issue where an extra dereference could
occur for a last-entered user being ejected if a user identifier was
also provided.

(closes issue ASTERISK-21907)
Reported by: Alex Epshteyn
Review: https://reviewboard.asterisk.org/r/2844/
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Merged revisions 399033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 399034 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 399035 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399036 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add a straightforward synopsis and description to the identify config object
in XML documentation.

(issue ASTERISK-22311)
(closes issue ASTERISK-22311)
Reported By: Rusty Newton
........

Merged revisions 399031 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399032 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These two commands must deal with the live bridges container for tab
completion and not the stasis cache.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399022 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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Merged revisions 399019 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399020 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 *  One bug fix. Made the synopsis for "type" to accurate.
 *  changing the usage of "IP-domains" to "IP addresses"
 *  clarifying the usage for the options, by adding a relevant description for
    each
 *  modified other areas of the XML help for clarity, such as the module
    description and a few synopsis changes here and there. See the patch.

(issue ASTERISK-22458)
(closes issue ASTERISK-22458)
Reported By: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2823/
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Merged revisions 399017 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399018 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-22424)
Reported by: Jonathan Rose
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399006 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The masquerade super test is failing on v12 with high fence violations and
crashing.  The fence violations are showing that party id allocated memory
strings are somehow getting corrupted in the
bridge_reconfigured_connected_line_update() function.  The invalid string
values happen to be the freed memory fill pattern.

After much puzzling, I deduced that the
bridge_reconfigured_connected_line_update() is copying a string out of the
source channel's caller party id struct just as another thread is updating
it with a new value.  The copying thread is using the old string pointer
being freed by the updating thread.  A search of the code found the
unreal_colp_redirect_indicate() routine updating the caller party id's
without holding the channel lock.

A latent bug in v1.8 and v11 hatched in v12 because of the bridging and
connected line changes.  :)

(issue ASTERISK-22221)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2839/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We shouldn't be exporting any symbols that start with pjsip_.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398928 65c4cc65-6c06-0410-ace0-fbb531ad65f3

You are adding dial strings to the queue, not channels. An aribitrary string
could be used, but you are typically referencing a channel. Correcting the
command help text.

(issue ASTERISK-22263)
(closes issue ASTERISK-22263)
Reported By: Rusty Newton
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Merged revisions 398884 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398887 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Changing text in chan_dahdi.conf sample to be accurate.

(issue ASTERISK-22308)
(closes issue ASTERISK-22308)
Reported By: Malcolm Davenport
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Merged revisions 398880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398883 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If we receive a 200 OK without SDP, we will now check to see if
the remote address has been established for that channel's RTP
session and if the to tag for that channel has changed from
the most recent to tag in a response less than 200.
If either a change has been made since the last to-tag was
received or the remote address is unset, then we will drop
the call.

(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header
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Merged revisions 398835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The denoise filter requires func_speex, not codec_speex.  Fix this in the
description of the denoise=yes option in confbridge.conf.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398822 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:

1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.

Also added an SDP when an update is sent out.

(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
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Merged revisions 398806 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398808 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are several locations in the code base where this is done:
buf = ast_realloc(buf, new_size);

This is going to leak the original buf contents if the realloc fails.

Review: https://reviewboard.asterisk.org/r/2832/
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Merged revisions 398757 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Race conditions between freeing a nul terminated string and
ast_strdup()'ing it are more likely to be detected if the fence and freed
magic numbers are completely different.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398732 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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Merged revisions 398694 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398695 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12.

With debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module list
(Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module
list will be locked before acquiring our mutex. In another thread, our
mutex will be locked before locking the module list (which happens in
the depths of calling backtrace()).

This patch fixes this issue by moving backtrace() calls outside of
critical sections that have the mutex acquired. The bigger change was to
reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed
that waiting on the mutex was equivalent to a single unlock (it actually
suspends all recursive locks on the mutex).

(closes issue ASTERISK-22455)
Review: https://reviewboard.asterisk.org/r/2824/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r398638 | dlee | 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line
  
  Added note about expected behavior of originate
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  r398639 | dlee | 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line
  
  Added note about expected behavior of originate (the rest of the commit)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398641 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
Multiple revisions 398559,398578

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  r398559 | kmoore | 2013-09-06 14:32:03 -0500 (Fri, 06 Sep 2013) | 20 lines
  
  Blocked revisions 398558
  
  ........
  Fix Jabber/XMPP distributed MWI
  
  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.
  
  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........
  
  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r398578 | kmoore | 2013-09-06 16:03:45 -0500 (Fri, 06 Sep 2013) | 1 line
  
  Unblock r398558


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a channel joins a multi-party bridge, the ordering of the CDRs that is
created is determined by the ordering of the channels who happen to be in that
bridge. When r398579 changed the number of buckets in the container to
something sensible, it changed the ordering that the CDRs was created in,
causing one of the multiparty tests to fail. This fixes the test with the
now expected ordering.
........

Merged revisions 398628 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Sometimes the Google Voice servers have a bad habit of sending out 1
byte replies to the xmpp resource. When a blank 1 byte reply is
received from the socket the buffer attempts to wait (endlessly) for
the rest of the reply from google which effectively blocks the socket
and google voice calls will no longer come into the server.

This patch allows the xmpp module to correctly detect empty packets and
send out ping replies to google. It also sets a socket timeout on the
default socket which prevents the xmpp socket from closing and
preventing future google voice calls from coming into the server.

Furthermore instead of sending an empty reply back to google we send a
proper xmpp ping reply back. This also adds several more
socket messages.

(closes issue ASTERISK-22347)
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/2771
Patches:
    xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524)
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Merged revisions 398618 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398620 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r398558 | kmoore | 2013-09-06 14:28:16 -0500 (Fri, 06 Sep 2013) | 17 lines
  
  Fix Jabber/XMPP distributed MWI
  
  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.
  
  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........
  
  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | 10 lines
  
  Commit the remainder of r398523
  
  This is a missing part of the commit in revision 398523 that corrects
  the name of a variable.
  
  (issue ASTERISK-22435)
  ........
  
  Merged revisions 398576 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix the temporary cdr candidate containers to use a prime number of
buckets.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398583 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix the LocalOptimizationBegin AMI event by eliminating an artificial
buffer size limitation that is too small anyway.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398574 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Added missing unregister of the cdr container in cdr_engine_shutdown().

* Fixed ref leak in off nominal path of cdr_object_alloc().

* Removed some unnecessary NULL checks in cdr_object_dtor().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398564 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed resulting warnings with improper use of ao2_global_obj_replace().

* Made a couple uses of ao2_global_obj_replace_unref(x, NULL) into the
equivalent and more appropriate ao2_global_obj_release() call.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When AST_DEVMODE is not defined, ast_asserts are not compiled into the
binary. In some cases, this means variables are not referenced or are
set but unused which causes warnings to show up.

(closes issue ASTERISK-22446)
Reported by: Jason Parker (qwell)
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Merged revisions 398521 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes the things in chan_h323 that were missed or ignored in the
great channel opaquification and gets chan_h323 back into a compiling
state.

(closes issue ASTERISK-22365)
Reported by: Dmitry Melekhov
Patches:
    chan_h323.patch uploaded by Dmitry Melekhov
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Merged revisions 398510 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Make ao2_bt() not use single char variable names.

* Fix ao2_bt() formatting.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Reduce indentation in __attempt_transmit().

* Don't update the static last error time variable every time in
__schedule_action() and socket_read().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix stray reference to idle_list in cleanup_thread_list().  This may be
the reason for the note in iax2_process_thread() about threads not being
removed from the task lists.

* Move cleanup_thread_list(&idle_list) to after the other lists are
cleaned up.
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Merged revisions 398416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix bridgecallno deadlock avoidance.  When doing deadlock avoidance, you
need to retest the status of values for each loop to see if you still need
the lock for bridgecallno.

* As a safety check, after acquiring the bridgecallno lock you should
check if iaxs[bridgecallno] is NULL just like the current callno checks.

* Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any
deferred frames to ensure that the iostate is IDLE when it is placed back
into the idle list.  defer_full_frame() tries to ensure
iax2_process_thread() wakes up to process the frame.
........

Merged revisions 398379 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398381 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Used some of Rusty's suggested language plus also included
more SIPesque descriptions of where the URIs are actually
used in an outgoing REGISTER.

(closes issue ASTERISK-22390)
reported by Rusty Newton
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398369 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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Merged revisions 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398304 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-22386)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22414)
Reported by: Corey Farrell
Patches:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 398285 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398286 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issues ASTERISK-22414)
Reported by: Corey Farrell
Patch:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 398281 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398283 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398284 65c4cc65-6c06-0410-ace0-fbb531ad65f3