jrose: Author Summary

chan_sip: Check the right channel's host address for permit/denydirectmedia

Prior to this patch, when checking the addresses for permitdirectmedia and
denydirectmedia, Asterisk would stupidly check the host address of the channel
permit/deny was specified for rendering the feature useless (or at least
redundant to turning on/off directmedia) (the approach in trunk is going to
be somewhat different)

Review: https://reviewboard.asterisk.org/r/1899/
........

Merged revisions 366547 from http://svn.asterisk.org/svn/asterisk/branches/1.8

chan_sip: Check the right channel's host address for permit/denydirectmedia

Prior to this patch, when checking the addresses for permitdirectmedia and
denydirectmedia, Asterisk would stupidly check the host address of the channel
permit/deny was specified for rendering the feature useless (or at least
redundant to turning on/off directmedia) (the approach in trunk is going to
be somewhat different)

Review: https://reviewboard.asterisk.org/r/1899/

Coverity Report: Fix issues for error type CHECKED_RETURN for core

(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366106 from http://svn.asterisk.org/svn/asterisk/branches/10

Coverity Report: Fix issues for error type CHECKED_RETURN for core

(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366106 from http://svn.asterisk.org/svn/asterisk/branches/10

Coverity Report: Fix issues for error type CHECKED_RETURN for core

(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Coverity Report: Fix issues for error type CHECKED_RETURN for core

(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/

Coverity Report: Fix issues for error type UNINIT in Core supported modules

(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/
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Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366049 from http://svn.asterisk.org/svn/asterisk/branches/10

Coverity Report: Fix issues for error type UNINIT in Core supported modules

(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/
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Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366049 from http://svn.asterisk.org/svn/asterisk/branches/10

Coverity Report: Fix issues for error type UNINIT in Core supported modules

(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/
........

Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Coverity Report: Fix issues for error type UNINIT in Core supported modules

(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/

chan_sip: Check the right channel's host address for permit/denydirectmedia

Prior to this patch, when checking the addresses for permitdirectmedia and
denydirectmedia, Asterisk would stupidly check the host address of the channel
permit/deny was specified for rendering the feature useless (or at least
redundant to turning on/off directmedia) (the approach in trunk is going to
be somewhat different)

Review: https://reviewboard.asterisk.org/r/1899/
........

Merged revisions 366547 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Block on frameout if the hardware has enough samples to complete a frame.

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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Merged revisions 365989 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 365990 from http://svn.asterisk.org/svn/asterisk/branches/10

Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8

Prior to this patch, in order to restore that behavior, a function would have
to be used on the QueueMember to make the ringinuse option do anything, which
is pretty unreasonable.

(closes issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1860/
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Merged revisions 361907 from http://svn.asterisk.org/svn/asterisk/branches/10

Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8

Prior to this patch, in order to restore that behavior, a function would have
to be used on the QueueMember to make the ringinuse option do anything, which
is pretty unreasonable.

(closes issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1860/

Fix some stuff involving calls to memcpy and memset

The important parts of the patch were already applied through other updates.

(closes issue ASTERISK-19445)
Reported by: Makoto Dei
Patches:
        memset-memcpy-length.patch uploaded by Makoto Dei (license 5027)

Make 'help devstate change' display properly (get rid of excess comma)

(closes ISSUE-ASTERISK-19444)
Reported by: Makoto Dei
Patches:
        devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027)
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Merged revisions 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Make the MeetMeAdmin N command (mute all nonadmins) not mute admins

(Closes Issue ASTERISK-19335)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1843/
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Merged revisions 361090 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Introducing the log message unique call identifiers feature

Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs  won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging

Review: https://reviewboard.asterisk.org/r/1823/

Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END

(closes issue ASTERISK-17940)
Reporter: Nic Colledge
Patches:
        features_18.patch uploaded by Nic Colledge (license 6245)
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Merged revisions 350501 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Adds Subscription-State header to notify with call completion. per RFC3265

(Closes issue ASTERISK-17953)
Reported by: George Konopacki
Patches:
        19400.patch uploaded by mmichelson (license 5049)
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Merged revisions 349482 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 349502 from http://svn.asterisk.org/svn/asterisk/branches/10

Coverity Report: Fix issues for error type UNINIT in Core supported modules

(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/
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Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366049 from http://svn.asterisk.org/svn/asterisk/branches/10

Make ForkCDR e option not set end time of the newly forked CDR log

Prior to this patch, ForkCDR's e option would immediately set the end time of the forked
CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time
being roughly the same as it's beginning time (which is in turn roughly the same as the
original's end time).

(closes issue ASTERISK-19164)
Reported by: Steve Davies
Patches:
        cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)

Replace GNU old-style field designator extensions to fix clang warnings

(issue ASTERISK-19540)
Reported by: Makoto Dei
Patches:
        clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
........
Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8

Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8
(closes issue ASTERISK-19540)

Updates config with bootstrap where I changed configure.ac in r360488

(issue ASTERISK-17842)
Reported by: Bryon Clark
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Merged revisions 360574 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 360575 from http://svn.asterisk.org/svn/asterisk/branches/10

Make transfer not ignore port information with SIP.

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

Adds a transfer callee on hangup option (like with Dial option F) to queues.

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/

Adds UPGRADE.txt notes to r357266 indicating changes to transport option

(issue ASTERISK-19352)
Reported by: jamicque
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Merged revisions 357356 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357357 from http://svn.asterisk.org/svn/asterisk/branches/10

Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12

(issue ASTERISK-19352)
reported by: jamicque
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Merged revisions 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357400 from http://svn.asterisk.org/svn/asterisk/branches/10

Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE

(issue ASTERISK-19352)
Reported by: jamicque
Patches:
        asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
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Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357497 from http://svn.asterisk.org/svn/asterisk/branches/10

Add additional character type types to supported data types for cdr_adaptive_odbc

The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
this patch adds those along with some other character types to the list of types
cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really
UTF8 characters as much as it was a failure to respond to the exact type that was
declared/in use on that database.

(closes issue ASTERISK-19334)
Reported By: Igor Nikolaev
Patches:
        cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236)
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Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357458 from http://svn.asterisk.org/svn/asterisk/branches/10

revision 357386 -- oops, accidentally made it 10.3 to 10.4 instead of 10.2 to 10.3

(issue ASTERISK-19352)
reported by: jamicque
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Merged revisions 357405 from http://svn.asterisk.org/svn/asterisk/branches/10

Changes transport option in sip.conf so that using multiple instances doesn't stack.

Prior to this patch, Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to simply use the transport
option specified last. Also, if no transport option is applied now, the default will
automatically be UDP.

(closes ASTERISK-19352)
Reported by: jamicque
Patches:
        asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
        issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674)
Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header
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Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Changes transport option in sip.conf so that using multiple instances doesn't stack.

Prior to this patch, Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to simply use the transport
option specified last. Also, if no transport option is applied now, the default will
automatically be UDP.

(closes ASTERISK-19352)
Reported by: jamicque
Patches:
        asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
        issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674)
Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header

Resolve duplicate label used in multiple priorities for the same extension.

Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.

(closes issue ASTERISK-18807)
Reported by: Kenneth Shumard
Patches:
        pbx.c.patch uploaded by Kenneth Shumard (License 5077)
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Merged revisions 346954 from http://svn.asterisk.org/svn/asterisk/branches/1.8