Commit Graph

4101 Commits

Author SHA1 Message Date
Matthew Jordan
ebcccf8997 Fix handling of negative return code when storing voicemails in ODBC storage
When storing a voicemail message using an ODBC connection to a database, the
voicemail message is first stored on disk.  The sound file associated with
the message is read into memory before being transmitted to the database.
When this occurs, a failure in the C library's lseek function would cause a
negative value to be passed to the mmap as the size of the memory map to
create.  This would almost certainly cause the creation of the memory map to
fail, resulting in the message being lost.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863
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Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362202 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 21:42:12 +00:00
Jonathan Rose
ba0f044bde 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)
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Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362084 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-13 16:12:17 +00:00
Jonathan Rose
c0b9fe8530 Send relative path named recordings to the meetme directory instead of sounds
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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Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362080 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-13 15:38:08 +00:00
Jonathan Rose
683eacb59a 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-11 17:20:08 +00:00
Matthew Jordan
aa21d4fc6b Fix memory leak when using MeetMeAdmin 'e' option with user specified
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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Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 20:32:52 +00:00
Kinsey Moore
a485f44022 Add missing newlines to CLI logging
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Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 18:19:03 +00:00
Russell Bryant
b2f7b0c649 Remove a few more files related to chan_usbradio and app_rpt.
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Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 15:50:18 +00:00
Kinsey Moore
51f0e5c53d Remove unnecessary error message in app_dial.c
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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Merged revisions 361329 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361330 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 13:32:34 +00:00
Jonathan Rose
fc45af331b Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined
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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Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361270 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-05 17:22:30 +00:00
Jonathan Rose
e96a59acfd 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)
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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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Merged revisions 361143 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 18:08:28 +00:00
Jonathan Rose
97b2fa8de1 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 13:51:45 +00:00
Kinsey Moore
93781fa161 Fix the display of documentation for Transfer
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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Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361041 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03 20:14:01 +00:00
Jonathan Rose
655a8d4420 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 20:01:20 +00:00
Jonathan Rose
d501c2ea2d undoing 360785 due to merging mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 19:59:30 +00:00
Jonathan Rose
bf994f0e04 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 19:54:35 +00:00
Terry Wilson
dd9405db05 Fix setting CDR variables in the hangup extension
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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Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-28 19:39:24 +00:00
Russell Bryant
0ec73946fa app_page: Fix a memory leak on every Page().
dial_list is a dynamically allocated array that is allocated at the beginning
of Page() based on how many devices will be dialed.  This was never being freed.
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Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 360364 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-24 03:11:43 +00:00
Russell Bryant
71a1541b0c app_jack: fix datastore memory leak in error handling path.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-24 03:03:20 +00:00
Jonathan Rose
c6979ff581 Adds F option to Bridge application
Similar to dial and queue F option.

(Closes issue ASTERISK-19282)
Reported by: To
Patches:
	bridge_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1825/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-22 21:25:22 +00:00
Kinsey Moore
6ff8f14865 Prevent Echo() from relaying control, null, and modem frames
Echo()'s description states that it echoes audio, video, and DTMF except for #
while it actually echoes any frame that it receives other than DTMF #.  This
was causing frame storms in the test suite in some circumstances where Echo()
was attached to both ends of a pair of local channels and control frames
were being periodically generated.  Echo()'s behavior and description have
been modifed so that it only echoes media and non-# DTMF frames.
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Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 360034 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-20 20:42:34 +00:00
Jonathan Rose
0399daaa2e Prevent chanspy from binding to zombie channels
This patch addresses a bug with chanspy on local channels which roughly 50% of the time
would create a situation where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never be able to hang up.

(closes issue ASTERISK-19493)
Reported by: lvl
Review: https://reviewboard.asterisk.org/r/1819/
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Merged revisions 359892 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359898 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-16 21:00:07 +00:00
Mark Michelson
827f2eae92 Revert the pre-dial addition.
The code may be just fine, but it had not received a "ship it!" on
review board yet.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-16 15:38:45 +00:00
Mark Murawki
c65b41f57a Add options PreDial options 'b' and 'B' to app_dial
* Added 'b' and 'B' options to Dial.  These options will allow you to run
  last-minute dialplan on the caller and callee channels while the Dial
  application is executing, but before the call is started.  For example you
  can use the 'b' option to run dialplan on the callee channel to get the name
  of the newly created channel right away.

Review: https://reviewboard.asterisk.org/r/1229/

(closes issue: ASTERISK-19548)
Reported by: Mark Murawski
Tested by: Mark Murawski, Stefan Schmidt



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-15 18:58:25 +00:00
Matthew Jordan
c61d49d5cc Fix remotely exploitable stack overrun in Milliwatt
Milliwatt is vulnerable to a remotely exploitable stack overrun when using
the 'o' option.  This occurs due to the milliwatt_generate function not
accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of
samples it can put in the output buffer.

This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET
when determining the maximum number of samples allowed.  Note that at no
point is remote code execution possible.  The data that is written into the
buffer is the pre-defined Milliwatt data, and not custom data.

(closes issue ASTERISK-19541)
Reported by: Russell Bryant
Tested by: Matt Jordan
Patches:
  milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283)
  Note that this patch was written by Russell, even though Matt uploaded it
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Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359694 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-15 18:55:54 +00:00
Richard Mudgett
e9703da1d5 Add missing connected line macro calls to initial dial for Dial and Queue apps.
The connected line interception macros do not get executed when the
outgoing channel is initially created and that channel's caller-id is
implicitly imported into the incoming channel's connected line data.  If
you are using the interception macros, you would expect that they get run
for every change to a channel's connected line information outside of
normal dialplan execution.

Review: https://reviewboard.asterisk.org/r/1817/
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Merged revisions 359609 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-15 18:32:22 +00:00
Russell Bryant
45205716d7 app_chanisavail: Fix use of uninitialized variable.
Ensure that status is set before it is used by resetting it during each loop
iteration.  This could have resulted in incorrect results from this app.
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Merged revisions 359486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359491 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14 23:29:32 +00:00
Richard Mudgett
2019a7e6b9 Fix Dial m and r options and forked calls generating warnings for voice frames.
When connected line support was added, the wait_for_answer() variable
single changed its meaning slightly.  Unfortunately, the places where
single was used did not necessarily get updated to reflect that change.
Also audio/video frames were sent to all forked calls when the endpoints
were never made compatible.

* Don't pass audio/video media frames when the channels have not been made
compatible.

* Added handling of AST_CONTROL_SRCCHANGE to app_dial.c.

* Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also
pass a requested MOH class.

(closes issue ASTERISK-16901)
Reported by: Chris Gentle

(closes issue ASTERISK-17541)
Reported by: clint

Review: https://reviewboard.asterisk.org/r/1805/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14 17:39:45 +00:00
Russell Bryant
6c9f009b6d Fix invalid reads/writes due to incorrect sizeof().
These few places in the code used sizeof() on h_addr in struct hostent.
This is sizeof(char *).  The correct way to get the size of this address is to
use h_length.  This error would result in reads/writes of 8 bytes instead of 4
on 64-bit machines.
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Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-14 10:05:07 +00:00
Russell Bryant
4585000039 Remove chan_usbradio and app_rpt.
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13 23:42:24 +00:00
Terry Wilson
7876521659 Fix IMAP storage compilation after opaquification changes
(closes issue ASTERISK-19513)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13 18:55:14 +00:00
Terry Wilson
786f5898d1 Finalize ast_channel opaquification
Review: https://reviewboard.asterisk.org/r/1786/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13 18:20:34 +00:00
Tilghman Lesher
9af5c769c3 Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4.
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13 08:06:20 +00:00
Joshua Colp
f5fda0eb74 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-10 20:06:46 +00:00
Sean Bright
4657b016ad Resolve a few more cases of variable shadowing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-08 17:48:14 +00:00
Sean Bright
99bd5b1e2e Eliminate a bunch of shadow warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-08 17:02:52 +00:00
Jonathan Rose
d3db6da254 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-02 16:57:12 +00:00
Richard Mudgett
2ce386d198 Remove bad usage of goto in ChanSpy next_channel().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-02 16:26:01 +00:00
Richard Mudgett
3af920cf01 Fix channel reference leak in ChanSpy.
* 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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Merged revisions 357809 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357810 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357815 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-02 16:03:38 +00:00
Terry Wilson
0e5c761c28 Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-01 22:09:18 +00:00
Walter Doekes
571cef491f Fix copying of CDR(accountcode) to local channels.
In r203638, during the addition of the Channel Event Logging, in mid-2009, this
got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the
CDR(accountcode) from the calling channel is available to dialed channels again
as well as showing up properly in the CDR's.

(closes issue ASTERISK-19384)
Reported by: jamicque
Patches: accountcode.patch (License #6033) by jamicque
Review: https://reviewboard.asterisk.org/r/1775/
Reviewed by: Richard Mudgett
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Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357576 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29 19:48:33 +00:00
Terry Wilson
a9d607a357 Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29 16:52:47 +00:00
Tilghman Lesher
d65de9c5c3 Correctly reset the dialplan priority.
When the stack frame is allocated, we save the address to which we should
return, when the Gosub returns.  However, if we just want to restore the
priority, then we need to subtract 1 before setting it.  Otherwise, when
a Gosub goes to a nonexistent address, it will skip a priority in the
dialplan.  This is because when we return from an application, the PBX
increments the priority for us.
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Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357421 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28 21:26:34 +00:00
Richard Mudgett
e063fa6b3f Fix REF_DEBUG compile errors.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28 20:34:11 +00:00
Richard Mudgett
39ffc96723 Remove dupliate 'i' option table entry in app_page.c.
(closes issue ASTERISK-19310)
Reported by: Makoto Dei
Patches:
      app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei
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Merged revisions 357352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357353 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28 19:37:04 +00:00
Kinsey Moore
1fac2fba4b Deprecated macro usage for connected line, redirecting, and CCSS
This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated.  This also adds
deprecation warnings for those features when used and in documentation.

Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27 16:50:19 +00:00
Matthew Jordan
5e40f2cd98 Fix crash in app_voicemail during close_mailbox
In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers.  However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL.  In that case, an invalid free would be attempted,
which could crash app_voicemail.  As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers.  This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
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Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356798 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-25 17:22:55 +00:00
Terry Wilson
ebaf59a656 Opaquification for ast_format structs in struct ast_channel
Review: https://reviewboard.asterisk.org/r/1770/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24 00:32:20 +00:00
Paul Belanger
26865092e6 Multiple revisions 356290,356335,356337
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  r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) | 4 lines
  
  Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
  
  Review: https://reviewboard.asterisk.org/r/1763/
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  r356335 | pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 lines
  
  Add back strsep() function for previous commit
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  r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb 2012) | 2 lines
  
  Missed one strsep() function
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Merged revisions 356290,356335,356337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356428 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-23 03:27:01 +00:00
Terry Wilson
57f42bd74f ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20 23:43:27 +00:00
Paul Belanger
73b5346e79 Fix channel opaquification for app_rpt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-16 22:00:31 +00:00