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
........

Merged revisions 377922 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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
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/
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
........

Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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/
........

Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix potential double free when unloading a module.
........

Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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)
........

Merged revisions 378119 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/
........

Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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)
........

Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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)
........

Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8
app_queue: Fix multiple calls to a queue member that is in only one queue.

When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.

* Fix so a queue member does not receive more than one call from a queue.

NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.

* Did some refactoring to eliminate some code redundancy.

(issue ASTERISK-16115)
Reported by: nik600
Patches:
      jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
      Modified

* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem.  The fix did not need to be optional.  The fix should not have
tried to explicitly set the device state.  Setting the device state by
something other than the device introduces a race condition.  I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
........

Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8
app_queue: Fix incorrect assertion.

(issue ASTERISK-16115)
Fix astcanary startup problem due to wrong pid value from before daemon call

When Asterisk forks itself into the background via a call to daemon, it must
re-set the pid value of the new process. Otherwise, astcanary gets the pid
value of the process before the fork, which prevents it from running. Asterisk
eventually starts lowering its priority, as it can no longer communicate
with the proverbial canary in the coal mine.

This patch ensures that the correct process identifier is used by astcanary.

Note that this is getting committed to 10 as a regression fix.

(closes issue ASTERISK-20947)
Reported by: Jakob Hirsch
Tested by: mjordan
patches:
  asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch (license 6113)
........

Merged revisions 379509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Update init.d scripts to handle stderr; readd splash screen for remote consoles

When r376428 was commited to re-order start up sequences to be more tolerant of
forking with thread primitives, a few items were changed that caused changes
in behavior on some distros. This includes:
 * Not displaying the splash screen on a remote console.
 * Displaying an error message on stderr when a remote console cannot connect
   to a running instance of Asterisk.

In the first case, the splash screen was re-added (thanks to Michael L. Young).
In the second case, the various init.d scripts were modified to pipe stderr
to /dev/null, as the error message is useful - if you execute a remote
console or a remote console command execution and it fail, it should tell
you. Note that the error message was always present, it just failed to be
printed prior to r376428.

Much thanks to the folks who quickly reported this problem, provided solutions,
and promptly tested the various init.d scripts on a variety of distros.

*** NOTE ***
If you're wondering why this got merged into 10, it's because we're going to
release a regression release with this in it. Might as well get it into the
branch so the tag reflects some version of reality.

(closes issue ASTERISK-20945)
Reported by: Warren Selby
Tested by: Michael L. Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan
patches:
  asterisk-20945-remote-intro-msg.diff uploaded by elguero (license 5026)
  ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan (license 6283)
........

Merged revisions 379760 from http://svn.asterisk.org/svn/asterisk/branches/1.8