Fix a potential deadlock in chan_sip during transfers.

The issue comes from the fact that transfers may perform
a redirecting update on a channel. The issue is that lock
inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process.

The fix is to move when the redirecting update occurs to a
place where neither the tech_pvt or the channel is locked so
that the two can be locked in the proper order.

(closes issue ASTERISK-20708)
reported by Mark Michelson
patches:
	ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)

Tested by:
	Tim Ringenbach at Asteria Solutions Group
Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases

See CHANGES-* files in English extra 1.4.12 tarballs for new sound prompts added.

(closes ASTERISK-20328)
Reported by: Matt Jordan
(closes AST-755)
Reported by: John Bigelow
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Merged revisions 377922 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 377923 from http://svn.asterisk.org/svn/asterisk/branches/10
Ensure Min-SE is included in outbound INVITEs

Asterisk now includes Min-SE in outbound INVITEs when the value is not
90 (the default) and session timers are not disabled. This has the
effect of Asterisk following RFC4028 more closely with regard to 422
responses and preventing situations in which Asterisk would be forced
to temporarily accept a call to tear it down based on a Session-Expires
below the locally configured Min-SE.

(issue SWP-5051)
Review: https://reviewboard.asterisk.org/r/2222/
Reported-by: Kinsey Moore
Patch-by: Kinsey Moore
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Merged revisions 377946 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 377947 from http://svn.asterisk.org/svn/asterisk/branches/10
Fix skinny debug tab completion

Review the syntax of the 'skinny debug' command to show more than
just 'show' for options to 'skinny debug' command.

(closes issue ASTERISK-20789)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
    skinny-debug.diff uploaded by snuffy (license 5024)
Minor fixes for chan_skinny

Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and 
correct len of 2 strcmp in skinny_setdebug(). (see opticron's review
on https://reviewboard.asterisk.org/r/2240/)
confbridge: Fix MOH on simultaneous user entry to a new conference.

When two users entered a new conference simultaneously, one of the callers
hears MOH.  This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.

* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code.  Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.

* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.

* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference.  This way any pre-join file playback does not
need to worry about MOH.

* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.

(closes issue ASTERISK-20606)
Reported by: Eugenia Belova
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2232/
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Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10
Fix skinny to recognise vmexten in general section of conf

Fixup the vmexten so if globally set in general section will be honored by
chan_skinny. Also get rid of the 'global_' part of variable name to match
regexten.

(closes issue ASTERISK-20790)
Reported by: snuffy
Tested by: snuffy, myself
Patches: 
    skinny-vm.diff uploaded by snuffy (license 5024)
app_queue: Revert bad ringinuse=no patch.

With the option ringinuse=no set, the patch committed for ASTERISK-16115
causes non-SIP queue members to never be called because the device state
is checked after a channel is created to determine if the member is busy.
These queue members always get the "Member %s is busy, cannot dial"
message.

Most channel drivers other than chan_sip use the default device state
handling.  The default device-state state is considered in use or unknown
if the channel exists or not respectively.

(closes issue ASTERISK-20801)
Reported by: rmudgett
Patches:
      jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett
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Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378037 from http://svn.asterisk.org/svn/asterisk/branches/10
Make libasteriskssl.so symlink use a relative path.

This was causing issues when using DESTDIR, since the path to which the link
pointed is not likely to exist (and not useful to exist) on the target system.

(issue ASTNOW-284)
Make chan_local module references tied to local_pvt lifetime.

The chan_local module references were manually tied to the existence of
the ;1 and ;2 channel links.

* Made chan_local module references tied to the existence of the local_pvt
structure as well as automatically take care of the module references.

* Tweaked the wording of the local_fixup() failure warning message to make
sense.

Review: https://reviewboard.asterisk.org/r/2181/
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Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378089 from http://svn.asterisk.org/svn/asterisk/branches/10
Fix potential double free when unloading a module.
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Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378093 from http://svn.asterisk.org/svn/asterisk/branches/10
Add test events for time limit-related hangups

This patch adds hangup-related test events in order to support testing
of time-limited bridges. This aids in testing the S() and L() bridge
options.

(issue SWP-4713)
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Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378120 from http://svn.asterisk.org/svn/asterisk/branches/10
Add branch-1.8-merged property to allow direct merging from v1.8
Give the causes[] a struct name.
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Merged revisions 378164 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Ensure chan_sip rejects encrypted streams without crypto info

This ensures that Asterisk rejects encrypted media streams (RTP/SAVP
audio and video) that are missing cryptographic keys and ensures that
the incoming SDP is consistent with RFC4568 as far as having a crypto
attribute present for any SAVP streams.

Review: https://reviewboard.asterisk.org/r/2204/
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Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378218 from http://svn.asterisk.org/svn/asterisk/branches/10
Resolve crashes due to large stack allocations when using TCP

Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.

This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
 * For SIP, the allocation now has an upper limit
 * For HTTP, the allocation is now a heap allocation instead of a stack
   allocation
 * For XMPP (in res_jabber), the allocation has been eliminated since it was
   unnecesary.

Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.

(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
  ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
  issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
  issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
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Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10
Prevent exhaustion of system resources through exploitation of event cache

Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.

This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.

(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
  event-cachability-3.diff uploaded by jcolp (license 5000)
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Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10
Restore branch-1.8-merged on 11

This was accidentally deleted during a merge.
Fix AMI redirect action with two channels failing to redirect both channels.

The AMI redirect action can fail to redirect two channels that are bridged
together.  There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.

* Made the bridge wait for both channels to be redirected before exiting.

* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.

* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding.  Previously the code fell back to a single channel
redirect operation.

(closes issue ASTERISK-18975)
Reported by: Ben Klang

(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
      jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode

Review: https://reviewboard.asterisk.org/r/2243/
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Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Prevent crashes from occurring when reading from data sources with large values

When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.

This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.

(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
 * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
 * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
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Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8