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

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358908 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358861 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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/



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

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358691 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358647 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358646 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358609 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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All of these were pretty obviously unused. Some were unused because
the code that used them was #if 0'd. In those cases, I just commented
out the unused-but-set variables.
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(from kpfleming's commit to trunk r356292)
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x=++x and x=x=1? Really?
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This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive.  This should also preserve the original case of
the device string as passed in to the event system.  CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.

The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.

This adds a unit test to verify that the event system works as expected.

(closes issue ASTERISK-19422)
Review: https://reviewboard.asterisk.org/r/1780/
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The check if an ISDN call is bridged before it could be placed on hold is
not necessary and is overly restrictive.  The check was originally done to
prevent problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application.  The ISDN transfer code has not required this restriction for
quite some time because ECT could transfer any two active calls to each
other.

* Remove ISDN hold restriction for calls connected to applications.

* Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.

(closes issue ASTERISK-19388)
Reported by: Birger Harzenetter
Tested by: rmudgett
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This should (and does in my testing) act just like the Dial option of the same name.
This allows a queue member to be transfered to the next priority (no args), or to
a context/extension/priority similar to goto (with args context^extension^priority)
when a caller hangs up on them.

(closes issue ASTERISK-19283)
Reported by: To
Patches:
	queue_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1785/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357861 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357821 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix next_channel() channel reference leak in ChanSpy.

(closes issue ASTERISK-19461)
Reported by: Irontec
Patches:
      app_chanspy_iteartor_next_unref.patch (license #6213) patch uploaded by Irontec

(issue ASTERISK-17515)
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