When you call the CONFIG dialplan function with the name of a variable that
doesn't exist in the target context you get an ERROR.  This does nothing but
clutter up the logs with messages that may be perfectly acceptable.  Just
because a variable wasn't in the context doesn't mean it's an error.  Maybei
t's optional or just needs to be defaulted or ignored.

This patch changes the log level from ERROR to DEBUG.  If a dialplan developer
wants to debug their dialplan they still canby setting the console debug level 
as needed.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3919/
........

Merged revisions 421327 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421328 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421329 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421337 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r421311 | mjordan | 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) | 9 lines
  
  res/ari/resource_channels: Don't return allocation failure on failed function
  
  If a function fails to execute, it is most likely due to one of two reasons:
  (1) The function doesn't exist or can't be read from
  (2) The function is dangerous and is restricted based on the user's permissions
  
  Currently we return allocation failure, which is incorrect. This updates the
  reason code to more accurately reflect why the request failed.

  ASTERISK-24215
........
  r421312 | mjordan | 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) | 4 lines
  
  res/ari/resource_channels: Fix compilation issue
  
  Forgot a parameter. Whoops.
........

Merged revisions 421311-421312 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421313 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch addresses a few issues:

1) The order of Dial events have been changed when performing a call forward.
   The order has now been altered to
    1) Dial begins dialing channel A.
    2) When A forwards the call to B, we issue the dial end event to channel
       A, indicating the dial is being canceled due to a forward to B.
    3) When the call to channel B occurs, we then issue a new dial begin to
       channel B.

2) Call forwards are now reported on the calling channel, not the peer channel.

3) AMI DialEnd events have been altered to display the extension the call is
   being forwarded to when relevant.

4) You can now get the values of channel variables for channels that are not
   currently in the Stasis application. This brings the retrieval of channel
   variables more in line with the rest of channel read operations since they
   may be performed on channels not in Stasis.

ASTERISK-24134 #close
Reported by Matt Jordan

ASTERISK-24138 #close
Reported by Matt Jordan

Patches:
	forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

Review: https://reviewboard.asterisk.org/r/3899
........

Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The same function, meetme_stasis_generate_msg, handles creating and publishing
Stasis message both when there are channels in the MeetMe conference and when
there are no channels in the conference. When the performance improvement was
made to use cached snapshots, this created a situation where Asterisk would
crash: obtaining a cached snapshot is not NULL tolerant.

This patch restores the previous implementation, which used a NULL safe set
of routines to produce a blob containing the channel snapshot (if available)
and information about the MeetMe conference.

ASTERISK-24234 #close
Reported by: Shaun Ruffell
Tested by: Shaun Ruffell
........

Merged revisions 421270 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421273 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The 'z' option is supposed to disable the dial timeout in the case of a call
forward. Unfortunately, the wrong timeout timer was passed to the do_forward
function, resulting in the option not working.

ASTERISK-24225 #close
Reported by: dimitripietro
Tested by: dimitripietro
patches:
  jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621)
  jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621)
........

Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421233 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421235 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421236 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
executed with optimization. This "help" unfortunately results in re-definition
warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.

Review: https://reviewboard.asterisk.org/r/3912/

ASTERISK-24032 #close
Reported by: Kilburn
Tested by: Kilburn, wdoekes
patches:
  1.8.diff uploaded by cloos (License 5956)
  10.diff uploaded by cloos (License 5956)
  11.diff uploaded by cloos (License 5956)
  12.diff uploaded by cloos (License 5956)
  13.diff uploaded by cloos (License 5956)
........

Merged revisions 421227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421228 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421229 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421230 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421231 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3914/
........

Merged revisions 421210 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421211 65c4cc65-6c06-0410-ace0-fbb531ad65f3

r420934 introduced some failures in the test suite.  Upon investigating, it was
discovered that differences in the way we were evaluating whether a channel was in
the process of leaving a bridge were causing some reinvites not to occur (mostly
reinvites back to Asterisk when ending a call). This patch fixes that behavioral
change.

ASTERISK-24027 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3910/
........

Merged revisions 421186 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421187 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421195 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.

Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
........

Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421164 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421165 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421166 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need
to be included, as the module does not using PJPROJECT any fashion.
Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as
a dependency, this also meant that res_hep_rtcp will fail to compile on a
system without PJPROJECT.

This patch removes the include.

Thanks to Damien Wedhorn for pointing this out in #asterisk-dev.

ASTERISK-24236 #close
Reported by: Damien Wedhorn, Matt Jordan
Tested by: Damien Wedhorn
........

Merged revisions 421064 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421065 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421066 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is being done in advance of the test for ASTERISK-23953
........

Merged revisions 421059 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421060 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421061 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421062 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CEL typically tracks a lot of information using the unique ID of the channel.
This is typically needed due to tying events together using the linked ID of
the various channels involved in a "call", which is derived from the channel ID
of the oldest channel involved in a bridge (or in the case of a Dial, the
parent channel).

Previously, we had updated the extra fields to include the involved channel
names, but forgot to put in the unique ID. This patch corrects that error.
........

Merged revisions 421037 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421042 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421043 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reduce the scope of local_peer and only get it if the ARI originate is
subscribing to the channels.

Review: https://reviewboard.asterisk.org/r/3905/
........

Merged revisions 421009 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421010 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421012 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Use ao2_replace() instead of ao2_cleanup(); ao2_bump().

ao2_replace() has the advantange of not altering the ref count if the
replaced pointer is the same.

Review: https://reviewboard.asterisk.org/r/3904/
........

Merged revisions 420992 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420993 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420956 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420957 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents a crash from occurring when a contact with no URI is used
for the creation of an outbound out-of-dialog request with no
associated endpoint.
........

Merged revisions 420949 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420950 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420953 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.

ASTERISK-24027 #close
Reported by: mjordan
Review: https://reviewboard.asterisk.org/r/3900/
........

Merged revisions 420934 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420940 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420919 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420921 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r399267, the verbose2magic stuff was edited. This time it results
in magic characters in the log files for multiline messages.

In trunk (and 13) this was fixed by the "stripping" of those
characters from multiline messages (in r414798).

This fix is altered to actually strip the characters and not replace
them with blanks.

Review: https://reviewboard.asterisk.org/r/3901/
Review: https://reviewboard.asterisk.org/r/3902/
........

Merged revisions 420897 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 420898 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420899 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Symptom is most likely an invalid ao2 object bad magic number message or a
less likely crash.
........

Merged revisions 420881 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420882 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made use ast_copy_string() instead of strcpy() for snoop uniqueid for
safety.  There is no guarantee that the max channel uniqueid length will
remain the same as the snoop uniqueid space.
........

Merged revisions 420879 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420880 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420856 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420836 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420837 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is to support the backwards compatible changes made in the next version
of Asterisk.
........

Merged revisions 420805 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420808 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420811 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Return the correct value instead of always returning 0 when setting
internal status on unreal channels.

Reported by: Richard Mudgett
........

Merged revisions 420802 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420803 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The patch to catch channels being shoehorned into Stasis() via external
mechanisms also happens to catch Announcer and Recorder channels
because they aren't known to be stasis-controlled channels in the usual
sense. This marks those channels as Stasis()-internal channels and
allows them directly into bridges.

Review: https://reviewboard.asterisk.org/r/3903/
........

Merged revisions 420795 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420796 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420797 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The unit tests require a sorcery.conf file that has been
set up to store resource lists in memory rather than retrieving
from configuration.

With a setup that is not conducive to running the tests, a fault
in sorcery currently causes Asterisk to crash when attempting to
run any of the tests.

To get around the crash, this adds a function that verifies the
current environment and marks the tests as "not run" if the setup
is not correct.
........

Merged revisions 420779 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420780 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Running testsuite tests locally produced no errors, but when
run using the continuous integration framework, crashes occurred.

The crashes occurred due to a refcounting error that had been fixed
for a similar situation.
........

Merged revisions 420758 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These modules were originally specified as being disabled, as they were
introduced midstream in Asterisk 12. That makes it nicer for folks who are
upgrading to a new release in the middle of Asterisk 12. That's not the case
for Asterisk 13: it's a brand new release. There's no reason to have the
modules disabled by default in that case.
........

Merged revisions 420742 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420743 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the space left in a stringfield is between 0 and
(alignof(ast_string_field_allocation)-1) adding new data would cause
memory corruption, because we would assume enough space (unsigned
underrun).

Thanks Arnd Schmitter for reporting and finding out the cause!

ASTERISK-23508 #close
Reported by: Arnd Schmitter
Tested by: Arnd Schmitter, JoshE

Review: https://reviewboard.asterisk.org/r/3898/
........

Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 420715 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 420716 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420717 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420718 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 420655 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 420656 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 420657 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420658 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch merely reformats and cleans up a bit of the jitterbuffer
documentation for the wiki.
........

Merged revisions 420639 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch gives the optional ability to keep queue rules in RealTime. It is
important to note that with this patch:
 (a) Queue rules in RealTime are only examined on module load/reload
 (b) Queue rules are loaded both from the queuerules.conf file as well as the
     RealTime backend
To inform app_queue to examine RealTime for queue rules, a new setting has been
added to queuerules.conf's general section "realtime_rules". RealTime queue
rules will only be used when this setting is set to "yes".

The schema for the database table supports a rule_name, time, min_penalty, and
max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or
'+' literal is provided. Otherwise, the penalties are treated as constants.

For example:
rule_name, time, min_penalty, max_penalty
'default', '10', '20', '30'
'test2', '20', '30', '55'
'test2', '25', '-11', '+1111'
'test2', '400', '112', '333'
'test3', '0', '4564', '46546'
'test_rule', '40', '15', '50'

which would result in :

Rule: default
 - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust
   QUEUE_MIN_PENALTY to 20
Rule: test2
 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
   QUEUE_MIN_PENALTY to 30
 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust
   QUEUE_MIN_PENALTY by -11
 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
   QUEUE_MIN_PENALTY to 112
Rule: test3
 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust
   QUEUE_MIN_PENALTY to 4564
Rule: test_rule
 - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust
   QUEUE_MIN_PENALTY to 15

If you use RealTime, the queue rules will be always reloaded on a module
reload, even if the underlying file did not change. With the option disabled,
the rules will only be reloaded if the file was modified.

Review: https://reviewboard.asterisk.org/r/3607/

ASTERISK-23823 #close
Reported by: Michael K
patches:
  app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621)
........

Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420609 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420592 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-24045
Reported by: Jacob Barber
Review: https://reviewboard.asterisk.org/r/3833/
........

Merged revisions 420577 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420562 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420538 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420539 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 420536 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420537 65c4cc65-6c06-0410-ace0-fbb531ad65f3