AST_PKG_CONFIG_CHECK: Similar to AST_EXT_LIB_CHECK, but simply uses
pkg-config data.

This simple version only uses pkg-config(1)'s tests.

This commit also uses the macro to test for GTK2 and GMIME (instead of
the current direct usage of pkg-config).

Review: https://reviewboard.asterisk.org/r/1906/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366351 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes a potential crash in mp3_read() by not assuming that
dbuf has enough data to finish filling up the output buffer.  The patch
also makes sure that the dbuf state gets reset after we know we read
everything out of it already.

In passing, this patch includes some other cleanups of this module,
including stripping trailing whitespace, formatting fixes based on
coding guidelines, and removing a number of unused members from the
private state struct.

(closes issue ASTERISK-19761)
Reported by: Chris Maciejewsk
Tested by: Chris Maciejewsk
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* Eliminate redundant list not empty check in clone_variables().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Before this patch, the predial routine executes on the ;1 channel of a
local channel pair.  Executing predial on the ;1 channel of a local
channel pair is of limited utility.  Any channel variables set by the
predial routine executing on the ;1 channel will not be available when the
local channel executes dialplan on the ;2 channel.

* Create ast_pre_call() and an associated pre_call() technology callback
to handle running the predial routine.  If a channel technology does not
provide the callback, the predial routine is simply run on the channel.

Review: https://reviewboard.asterisk.org/r/1903/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366183 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20,
22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved.  Those skipped
were either extended/deprecated or in areas of code that shouldn't be
disturbed.

(Closes issue ASTERISK-19650)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366169 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366126 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue AST-998)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366062 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366051 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes some problems with skipping audio in elaborate scenarios involving
multiple codecs by making codec_dahdi operate in a more synchronous
fashion similar to codec_g729. This change also fixes the use of file
conversion tools from Asterisk's CLI. This change may cause the thread
responsible for transcoding audio to block briefly (Shaun Ruffell describes
this as 'several milliseconds') while waiting for the hardware transcoder.

(closes issue ASTERISK-19643)
reported by: Shaun Ruffell
Patches:
	0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
	uploaded by Shaun Ruffell (license 5417)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow menuselect to get its set of CFLAGS and LDFLAGS through the
environment of Make:

  make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"

Review: https://reviewboard.asterisk.org/r/1907/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366002 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If you hit the wrong DTMF digit trying to accept/decline a FollowMe call,
you had to wait for the prompt to repeat to try again.

* Make FollowMe compare the last DTMF digits received to the
accept/decline matching strings.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_sip was coded under the assumption that a SIP dialog with an owner channel
will always be destroyed after the owner channel has been hung up.

However, there are situations where the SIP dialog can time out and auto destruct
before the corresponding channel has hung up. A typical example of this would be
if the 'h' extension in the dialplan takes a long time to complete. In such cases,
__sip_autodestruct() would complain about the dialog being auto destroyed with
an owner channel still in place. The problem is that even once the owner channel
was hung up, the sip_pvt would still be linked in its ao2_container because nothing
would ever unlink it.

The fix for this is that if __sip_autodestruct() is called for a sip_pvt that still
has an owner channel in place, the destruction is rescheduled for 10 seconds in the
future. This will continue until the owner channel is finally hung up.

(closes issue ASTERISK-19425)
reported by David Cunningham
Patches:
    ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)

(closes issue ASTERISK-19455)
reported by Dean Vesvuio
Tested by Dean Vesvuio
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The FollowMe caller call leg is usually answered and listening to MOH.
The caller could put the call on hold while FollowMe is looking for a
winner.  The winning outgoing call is now immediately placed on hold if
the caller has put the call on hold before the winning call was selected.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365829 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Why this tiny struct was malloced instead of the 28k struct in the last
change is beyond me.  Just doing my part to help stamp out sillyness.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Sending the 'I' command from an external process will cause the current playlist
to be cleared, including stopping any audio file that is currently playing.  This
is useful when you want to interrupt audio playback only when specific DTMF is
entered by the caller.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Helping to stamp out stack abuse.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix FollowMe leaving recorded caller name file on error paths in
app_exec().

* Use correct buffer dimension define in struct fm_args.namerecloc[].
This fixes unexpected namerecloc filename length restriction.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made use MAX_YN_STRING define to make all accept/decline DTMF buffers
the same size.  Just using 20 isn't good enough when someone didn't get
the memo.

* Fix stupid use of a global variable in FollowMe.  (ynlongest)

* Fix bit field declarations in FollowMe.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365633 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This uses the calling channel's caller ID and connected line information
to populate the remote and local identities in the dialog-info NOTIFY when
an extension is ringing.

There is a bit of an oddity here, and that is that we seed the remote target
with the To header of the outbound call rather than the from header. This
is because it was reported that seeding with the from header caused hints
to be broken with certain SNOM devices. A comment has been added to the code
to explain this.

(closes issue ASTERISK-16735)
reported by Maciej Krajewski
patches:
    local_remote_hint2.diff uploaded by Mark Michelson (license #5049)
	16735_tweak1.diff uploaded by Mark Michelson (license #5049)
Tested by Niccolo Belli
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365532 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Those channels are opaque now...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365480 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The VoiceMail d([c]) option is documented to accept digits for a new extension
in context <c>, if played during the greeting.  This option works fine if the
extension being redirected to has an extension with the same initial digit in
the channel's current context.  If that digit did not happen to exist in some
extension, a dialplan match would fail and the user would not be redirected.

This patch fixes it such that if the <c> option is used, the extensions are
matched in that context as opposed to the caller's original context.

(closes issue ASTERISK-18243)
Reported by: mjordan
Tested by: mjordan

Review: https://reviewboard.asterisk.org/r/1892
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365477 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Most of the changes here are trivial NULL checks.  There are a couple
optimizations to remove the need to check for NULL and outboundproxy parsing
in chan_sip.c was rewritten to avoid use of strtok.  Additionally, a bug was
found and fixed with the parsing of outboundproxy when "outboundproxy=," was
set.

(Closes issue ASTERISK-19654)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made chan_local.c:check_bridge() check the return value of
ast_channel_masquerade().  In long chains of local channels, the
masquerade occasionally fails to get setup because there is another
masquerade already setup on an adjacent local channel in the chain.

* Made the outgoing local channel (the ;2 channel) flush one voice or
video frame per optimization attempt.

* Made sure that the outgoing local channel also does not have any frames
in its queue before the masquerade.

* Made do the masquerade immediately to minimize the chance that the
outgoing channel queue does not get any new frames added and thus
unconditionally flushed.

* Made block indication -1 (Stop tones) event when the local channel is
going to optimize itself out.  When the call is answered, a chain of local
channels pass down a -1 indication for each bridge.  This blizzard of -1
events really slows down the optimization process.

(closes issue ASTERISK-16711)
Reported by: Alec Davis
Tested by: rmudgett, Alec Davis
Review: https://reviewboard.asterisk.org/r/1894/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365356 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These three all are in RTP code that attempts to print the number of sequence number cycles
in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number
right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without
the bit masking.

(issue ASTERISK-19649)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The security events framework API was changed in Asterisk 10 but the unit tests
were not updated at the same time.

This patch does the following:
* Adds two more security events that were added to the API 
* Add challenge, received_challenge and received_hash in the inval_password 
  security event unit test

(Closes issue ASTERISK-19760)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
issue-asterisk-19760-trunk.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1897/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The current CHANGES file refers to doc/ in many places and those files no longer exist.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

from value to pointer per functions that use this.

(close issue ASTERISK-19670)
Reported by: Matt Jordan
Patches:
  ASTERISK-19670.patch (License #5415)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365161 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1896/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

instead of without data buffer

(close issue ASTERISK-19674)
Reported by: Matt Jordan
Patches:
  ASTERISK-19674.patch (License #5415)
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........
  r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines
  
  Fix a CEL LINKEDID_END race and local channel linkedids
  
  This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes
  the race condition by no longer scanning the channel list for "other" channels
  with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings
  and uses the refcount of the string as a counter of how many channels with the
  linkedid exist. Not only does this eliminate the race condition, but it also
  allows us to look up the linkedid by the hashed key instead of traversing the
  entire channel list.
  
  Review: https://reviewboard.asterisk.org/r/1895/
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  r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines
  
  Don't leak a ref if out of memory and can't link the linkedid
  
  If the ao2_link fails, we are most likely out of memory and bad things
  are going to happen. Before those bad things happen, make sure to clean
  up the linkedid references.
  
  This patch also adds a comment explaining why linkedid can't be passed
  to both local channel allocations and combines two ao2_ref calls into 1.
  
  Review: https://reviewboard.asterisk.org/r/1895/
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........
Update security events unit tests

The security events framework API was changed in Asterisk 10 but the unit tests
were not updated at the same time.

This patch does the following:
* Adds two more security events that were added to the API 
* Add challenge, received_challenge and received_hash in the inval_password 
  security event unit test

(issue ASTERISK-19760)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1877/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365016 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is useful in cases where chan_sip may be listening on multiple addresses.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In audiohook_read_frame_both, anytime samples are obtained from the read/write
factories a debug statement is logged stating that samples were not obtained
from the factories.  This statement used to only occur if option_debug was
turned on and no samples were obtained; in some refactoring when the
option_debug statement was removed, the "else" clause was removed as well.

This patch makes it so that those debug log statements only occur if the
condition leading up to them actually happened.
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