Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

When merging in the patch for ASTERISK-22728, the UPGRADE.txt file was changed
incorrectly.  That change should have gone into ASTERISK-11.txt.

This commit is to fix that.

Also, another comment in the UPGRADE-11.txt was missing and this commit adds
that as well.
........

Merged revisions 402115 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402117 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(related to issue ASTERISK-21903)
........

Merged revisions 396199 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396200 65c4cc65-6c06-0410-ace0-fbb531ad65f3

indicate when streaming an audio file fails like it is done in other parts
of the code to indicate an error.

Note was requested by Paul Belanger: 
http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html

(related to issue ASTERISK-21903)
........

Merged revisions 396196 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 396197 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396198 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting.  Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.

This patch works similar to what occurs in build_peer().  We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.

In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.

This issue is also present in Asterisk 1.8 and a separate patch will be applied
to it.

(issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
  asterisk-21225-handle-options-default-prob_v4.diff
						Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2385/
........

Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This allows Asterisk to start without having to specify the
LD_LIBRARY_PATH. This can be disabled by passing --disable-rpath to
configure.

(closes issue ASTERISK-20407)
Reported by: David M. Lee
Review: https://reviewboard.asterisk.org/r/2132/
........

Merged revisions 379475 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379477 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371170 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370382 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370354 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.

(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif.

This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either.
These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold,
unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications.
The original Google Talk protocol is also supported for Google Voice interoperability.

You may ask yourself though where the name motif comes from... and I would say to you... music!

motif: a perceivable or salient recurring fragment or succession of notes

Sorta like a jingle!

Review: https://reviewboard.asterisk.org/r/1917/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ANI2 information is now parsed out of SIP From headers when present in
the oli, isup-oli, and ss7-oli parameters and is available via the
CALLERID(ani2) dialplan function.

(closes issue ASTERISK-19912)
Patch-by: Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1947/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368784 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-19322)

(closes issue ASTERISK-19875)
Reported by: call
........

Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368472 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.

This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.

Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.

* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.

Review: https://reviewboard.asterisk.org/r/1829/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in
favor of the Gosub method without a means of backwards compatibility.

(issue ASTERISK-18809)
(closes issue ASTERISK-19457)
Reported by: Matt Jordan
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1855/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The PeerStatus event for IAX2 channels currently includes a header named Post
which should have been Port.  Post was removed and the AMI version has been
updated to 1.3.
........

Merged revisions 359982 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359983 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality:
 * Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls.
 * Added ability for translation on-screen menu to multiple languages. Tested on Russian languages.  Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone
 * Other described in CHANGES file

Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa. 
Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E.

(closes issue ASTERISK-16890)

Review: https://reviewboard.asterisk.org/r/1243/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated.  This also adds
deprecation warnings for those features when used and in documentation.

Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is.  The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.

Review:  https://reviewboard.asterisk.org/r/1599/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-17053)
Reported by: justdave


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354084 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch removes res_ais and introduces a new module, res_corosync.
The OpenAIS project is deprecated and is now just a wrapper around
Corosync.  This module provides the same functionality using the same
core infrastructure, but without the use of the deprecated components.

Technically res_ais could have been used with an AIS implementation other
than OpenAIS, but that is the only one I know of that was ever used.

Review: https://reviewboard.asterisk.org/r/1700/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk now supports using ODBC with databases where a single schema must be
selected.  Previously, INSERTs would fail because they did not take into
account extra fields cause by having multiple schemas.  This also corrects
some SQL resource leaks.

(closes issue ASTERISK-17106)
Patch-by: Alexander Frolkin
Patch-by: Tilgnman Lesher


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353964 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add an AMI event in the Call category that is issued when a call is terminated
due to either RTP stream inactivity or SIP session timer expiration.

Event description:

Event: SessionTimeout
Source: source
Channel: channel-name
Uniqueid: channel-unique-id

`source` can be either RTPTimeout or SIPSessionTimer

(closes issue ASTERISK-16467)
Patch-by: Kirill Katsnelson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Adds per-parking lot options comebackcontext and comebackdialtime
* Makes comebacktoorigin settable per parking lot
* Sets a PARKER channel variable when comebacktoorigin is disabled

(closes issue ASTERISK-16643)
Reported by: Mitch Sharp (bluecrow76)
Patches:
asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231
with updates by me.

Review: https://reviewboard.asterisk.org/r/1674
Review: https://reviewboard.asterisk.org/r/963
Reviewed by Richard Mudgett



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Users created by users.conf with hasvoicemail=yes have been documented as
using a Gosub to stdexten since v1.6.0.  However, the code still generates
dialplan to access stdexten as a Macro as documented in v1.4; which does
not work with the newer extensions.conf.sample file.

* Make generated dialplan access the stdexten dialplan with the documented
Gosub instead of the older Macro style.

(closes issue ASTERISK-18809)
Reported by: Jay Allen
Patches:
      gosub_patch-pbx_config.patch (license #6323) patch uploaded by Jay Allen (modified)
Tested by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349782 65c4cc65-6c06-0410-ace0-fbb531ad65f3

During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655).  This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.

I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.

(issue ASTERISK-18974)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname,
language, locale, pager, password, tz.

(closes issue ASTERISK-18634)
Patch by: Kris Shaw
Review: https://reviewboard.asterisk.org/r/1568
Reviewed by: Walter Doekes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1599


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The udptl port range was defined as 4000-4999 in the udptl.conf.sample,
as 4500-4599 if you didn't have a config and 4500-4999 if your config
was broken. Default is now 4000-4999.

(closes issue ASTERISK-16250)
Reviewed by: Tilghman Lesher

Review: https://reviewboard.asterisk.org/r/1565
........

Merged revisions 343580 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AST-654)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/10

........
  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/
........
  r340220 | twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
  
  Add a missing file for the astdb2bdb conversion utility
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340221 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1434/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1427/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1432/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335555 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1429/

(closes issue ASTERISK-18497)

Thanks to russellb for peer review.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1426/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335170 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/10

........
  r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug 2011) | 2 lines
  
  Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES

  AST-580
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332033 65c4cc65-6c06-0410-ace0-fbb531ad65f3