This code just needed sizeof(int), not sizeof(int *).
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In this case, just remove the memset().  There was a redundant memset that is
done correctly just 2 lines later.
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This didn't actually result in a bug anywhere, luckily.  The only place
where the result of these memcpys was used is in app_dial, and the only
field that it read out of ast_call_feature was the first one, which is an
int, so these memcpys always copied just enough to avoid a problem.
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's' is set to 'workspace'.  Make sure 'workspace' doesn't go out of scope while
the reference to it via 's' is still used.
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Dump cache of published events when a node joins the cluster.

Also use a more reliable method for stopping the poll() thread.
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These modules are being maintained outside of the tree and have been for a long
time now, so it doesn't make sense to keep them here.

Review: https://reviewboard.asterisk.org/r/1764/
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...again.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added new chan_dahdi.conf colp_send option parameter to block connected
line updates per span.

(closes issue ASTERISK-17025)
Reported by: Michael Smith


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A previous CDR fix for setting CDR variables during a bridge via
custom dialplan features broke setting CDR variables in the
hangup extension. This patch fixes the issue.

Review: https://reviewboard.asterisk.org/r/1794/
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This patch drastically simplifies the device state aggegation code.
The old method was not only overly complex, but also made it impossible
to return AST_DEVICE_INVALID from the aggregation code. The unit test
update is as a result of fixing that bug.

The SIP change stems from a bug introduced by removing a DNS lookup
for hostname-based SIP channels.

(closes issue ASTERISK-16702)
Review: https://reviewboard.asterisk.org/r/1808/
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(closes issue ASTERISK-19513)


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Review: https://reviewboard.asterisk.org/r/1786/


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The set_format() function was more subtle in how it modified the
struct ast_channel readtrans/writetrans values.

* Fixed ast_activate_generator() conversion correctly.

(closes issue ASTERISK-19434)
Reported by: Birger Harzenetter
Tested by: rmudgett


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This change restores functionality that was present in 1.4, when AEL macros
were implemented with the Macro dialplan application.  Macros are fraught with
functionality issues, because they consume a large portion of the underlying
application stack.  This limits the ability of AEL users to call many layers
of subroutines, an issue which Gosub does not have (originally tested to
100,000 levels deep).  Therefore, starting in 1.6.0, AEL macros were
implemented with Gosub.

However, there were some implicit behaviors of Macro, which were not replicated
at the same time as with the transition to Gosub, one of which is documented in
the related issue.  In particular, the "h" extension is designed to execute not
in the Macro context, but in the topmost calling context.  Due to legacy issues
with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks
in all calling contexts, bubbling up from the deepest level until it finds an
"h" extension.

Since AEL hides the complexity of the underlying dialplan logic from the AEL
programmer, it's reasonable to assume that this behavior should not change in
the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break
working AEL configurations in the transition to Asterisk 1.8 LTS.  This fix
is the result, which implements a search for the "h" extension in all calling
Gosub contexts.

Fixes ASTERISK-19336

Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher
	(with slight modifications for 1.8)

Tested by: Johan Wilfer

Review: https://reviewboard.asterisk.org/r/1776/
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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/



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Transition app_page to using app_confbridge internally for the conference bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles.

Review: https://reviewboard.asterisk.org/r/1754/


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Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail
because port would be cut from the host string and ignored. This simply keeps chan_sip
from cutting off the port number during these kinds of transfers.

(closes issue ASTERISK-19321)
Reported by: Federico Alves
Review: https://reviewboard.asterisk.org/r/1790/diff/#index_header
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Passing -Wshadow to gcc enables shadow warnings.  From the gcc manual:

    Warn whenever a local variable or type declaration shadows another
    variable, parameter, type, or class member (in C++), or whenever a
    built-in function is shadowed.

Asterisk will not currently compile with this option set, but a number of bugs
have been discovered by enabling this flag on specific files.  The long-term
goal is to eliminate all of the suspect code that causes this warning to be
emitted.


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Patch also implements correct handling of datetime2 and datetimeoffset new
datatypes in SQL Server 2008 and 2008 R2.

(closes issue ASTERISK-17548)

Review: https://reviewboard.asterisk.org/r/1160/
Review: https://reviewboard.asterisk.org/r/1804/


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Update to:
(issue ASTERISK-19372)
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If the wctc4xxp returns more than a single packet, we need to update the number
of samples in the returned frame accordingly.

Acked-by: Shaun Ruffell <sruffell@digium.com>
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Without detecting these types, cel_odbc blows up when the character
set for the table is utf8. This also wraps cdr_adaptive_odbc's use of
those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the
code.
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* Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence().

Thanks to Sean Bright for compiling with -Wshadow and finding this bug.
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For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered.  If dialtone is
detected, the call is hung up.
no:       Disabled. (Default)
yes:      Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always:   Look for dialtone for the entire call.  Dialtone may return
          if the far end hangs up first.

dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always

(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper

Review: https://reviewboard.asterisk.org/r/1737/


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SS7 is a trunk protocol and should clear a failed call as soon as
possible.

* Made SS7 hangup a call immediately if it has not connected yet for
INCOMPLETE/BUSY/CONGESTION causes.  Otherwise, play an appropriate inband
tone.

(closes issue ASTERISK-19372)
Reported by: Igor Nikolaev
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Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel
that requires out-of-band DTMF will not work.

* Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and shows that
the code really did something useful.

* Improved some chan_dahdi DTMF debug messages to help track DTMF
handling.

(closes issue ASTERISK-19312)
Reported by: Igor Nikolaev
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The process_output function in manager.c attempted to call fclose and close immediately
afterwards. Since fclose implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error checking to fclose and
close depending on which was deemed necessary. Also error messages. Thanks to Rosen
Iliev for pointing out the location of the problem.

(closes issue ASTERISK-18453)
Reported By: Jaco Kroon
Review: https://reviewboard.asterisk.org/r/1793/
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(issue ASTERISK-19355)
Reported by: tomaso

(closes issue AST-825)
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Asterisk was not setting pendinginvite in the upper half of
handle_request_invite such that the 4xx was retransmitted repeatedly even
though an ack was received for every retransmission.

(closes issue ASTERISK-19303)
Reported by: Jon Tsiros
Patches:
  fix-19303.patch uploaded by Jeremiah Gowdy (license 6358)

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