For the formats that support seek and/or truncate operations, many of
the C library calls used to determine or set the current position indicator
in the file stream were not being checked.  In some situations, if an error 
occurred, a negative value would be returned from the library call.  This
could then be interpreted inappropriately as positional data.

This patch checks the return values from these library calls before
using them in subsequent operations.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362153 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362085 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, no effort was made to parse the path name to determine a proper
destination for recordings of MeetMe's r option. This fixes that.

Review: https://reviewboard.asterisk.org/r/1846/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362081 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This helps clean up the Asterisk CLI by converting the log message from verbose
to debug


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362043 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in
favor of the Gosub method without a means of backwards compatibility.

(issue ASTERISK-18809)
(closes issue ASTERISK-19457)
Reported by: Matt Jordan
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1855/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously, setting trunkfreq had no effect on initial load or on reload and
only ever used the default value.  This causes trunkfreq to be used 
appropriately on initial load and reload.

(closes issue ASTERISK-19521)
Patch-by: Jaco Kroon
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361987 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change to the build system rips out any usage of PROC along with
architecture-specific optimizations in favor of using -march=native where it is
supported.  This fixes broken builds on 64bit Intel systems and results in
better optimized code on systems running GCC 4.2+.

Review: https://reviewboard.asterisk.org/r/1852/
(closes issue ASTERISK-19462)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361968 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361909 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated.  If a new DAHDI channel is successfully created,
the event is passed up to the analog_ss_thread without error and the loop
exits.  If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level.  This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.

* Rework the -r361705 patch to better manage the cs and mtd allocated
resources.

* Fixed use of mwimonitoractive flag to be correct if the mwi_thread()
fails to start.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When unlinking itself from the registered HTTP URIs, res_http_post could
inadvertently free all URIs registered with the HTTP server.  This patch
modifies the unregister method to only free the URI that is actually
being unregistered, as opposed to all of them.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the global_curl_info data structure could not be allocated, the
datastore associated with the operation would be free'd, but the function
would not return.  This would later dereference the datastore, almost
certainly causing Asterisk to crash.  With this patch, if the data
structure is not allocated the method will return an error code, and
not attempt any further operation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated.  If a new DAHDI channel is successfully created, 
the event is passed up to the analog_ss_thread without error and the loop
exits.  If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level.  This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.

This patch makes it so that we only free the caller ID structure if a
DAHDI channel is successfully created, and we bump the gains back up
if we fail to make a DAHDI channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the SHARED function modifies a variable, it removes it from its list of
variables and reinserts the new value at the head of the list of variables.
Doing this inside a standard list traversal can be dangerous, as the
standard list traversal does not account for the list being changed.  While
the code in question should not cause a use after free violation due to its
breaking out of the loop after freeing the variable, it could lead to a
maintenance issue if the loop was modified.  This also fixes a violation
reported by a static analysis tool, which also makes this code easier to
maintain in the future. 
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361659 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the XML calendar data returned by a Microsoft Exchange Web Service
specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address
is provided, a condition existed where an ast_calendar_attendee struct would
be allocated but not appended to the list of attendees.  Because of that,
the memory associated with the attendee would never be freed.  This patch
frees the memory if no e-mail address is provided.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command
(eject last user that joined) is used in conjunction with a specified user.
Regardless of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user.  Because the 'e' option kicks
the last user that joined, as opposed to the one specified, the reference to
the user specified by the command would be leaked when the user variable
was assigned to the last user that joined.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361561 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key
header did not already exist.  If it already existed it would delete it.

* Made msg_set_var_full() exit early if the named variable did not already
exist and the value to set is empty.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361523 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines
  
  Fix typo in svn:keywords
........
  r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines
  
  Fix typo in svn:keywords
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361429 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361382 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added a '\n' to the warning messages when we ignore a media stream due to the
port number being '0'.

(closes issue ASTERISK-19646)
Reported by: Badalian Vyacheslav
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361334 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The error message for failure to stop autoservice after a gosub or macro call
during a dial was removed for macro while Asterisk 1.4 was still being actively
developed. The corresponding gosub error message was never removed.

(closes issue ASTERISK-19551)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There were a few instances of restarting music on hold in meetme that would cause
Asterisk to revert to the default class of music on hold for no adequate reason.

Review: https://reviewboard.asterisk.org/r/1844/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)
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Merged revisions 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361209 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361155 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Closes Issue ASTERISK-19335)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1843/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This came up while fixing documentation generation for many other cases where
the argument separator was not being displayed properly.  Now that it is
displayed properly, it shows up in the wrong place for Transfer since the '/'
is only required if Tech is present.

(related to issue ASTERISK-18168)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(related to ASTERISK-19575)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361039 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Hangup now can take a regular expression as the Channel option.  If you want
  to hangup multiple channels, use /regex/ as the Channel option.  Existing
  behavior to hanging up a single channel is unchanged, but if you pass a regex,
  the manager will send you a list of channels back that were hung up.

(closes issue ASTERISK-19575)
Reported by: Mark Murawski
Tested by: Mark Murawski



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change prevents Asterisk from sending RTCP receiver reports during a
remote bridge since it is no longer receiving media and should not be
reporting anything.

(related to ASTERISK-19366)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360994 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The logger_thread() had an exit path that failed to release the logmsgs
list lock.

* Make logger_thread() exit path unlock the logmsgs list lock.

* Made ast_log() not queue any messages to the logmsgs list if the
close_logger_thread flag is set.

(issue ASTERISK-19463)
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360935 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, a connected line update was queued during
call pickup and then an answer frame was queued. The original
caller would presumably then have his connected line updated
and then the call would be answered.

In actuality, the answer frame was not how the call ended up
being answered. Rather, an odd section in app_dial that checks
if the called channel's state is up.

The result is that the order of the connected line update and
the answer were variable. In most cases, this wasn't actually
a bad thing. However, if the 'I' option was passed to dial, the
connected line update would be inhibited.

The fix is to queued the connected line after the answer frame is
queued. This way the race in app_dial is between two
conditions resulting in an answer. This way the connected line
update occurs after the answer every time.

(closes issue ASTERISK-19183)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
    Mark Michelson
Patches:
    ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
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Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360886 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change makes use of connected party information in addition to caller ID in order
to populate local and remote XML elements in the dialog-info NOTIFYs.

(closes issue ASTERISK-16735)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
Patches:
    local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
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Merged revisions 360862 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360872 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Rename astobj2 API parameter funcname to func.

* Rename astobj2 API iterator parameter to iter.

* Update some documentation for OBJ_MULTIPLE.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I was getting confused during some testing why Asterisk was saying that
a subscription was being added when it was clearly being removed. This
fixes that confusion.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360673 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Global ao2 objects must always exist after initialization because there is
no access control to obtain another reference to the global object.

It is expected that module configuration could use these new API calls to
replace an active configuration parameter object with an updated
configuration parameter object.

With these new API calls, the global object could be replaced, removed, or
referenced without the risk of someone using a stale global object
pointer.

Review: https://reviewboard.asterisk.org/r/1824/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3