Age | Commit message (Collapse) | Author |
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* 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
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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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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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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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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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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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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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(closes issue ASTERISK-19513)
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Review: https://reviewboard.asterisk.org/r/1786/
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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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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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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/
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* 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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Review: https://reviewboard.asterisk.org/r/1784/
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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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Review: https://reviewboard.asterisk.org/r/1773/
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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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(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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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)
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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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Review: https://reviewboard.asterisk.org/r/1770/
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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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Review: https://reviewboard.asterisk.org/r/1753/
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Review: https://reviewboard.asterisk.org/r/1733/
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(closes issue ASTERISK-19188)
Reported by: slesru
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open_mailbox() was changed quite a long time ago to allocate this memory.
close_mailbox() should have been changed to be responsible for freeing it.
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This simply fixes the compilation issue introduced in r354429 by
re-adding the 'quote' variable.
(closes issue ASTERISK-19337)
Reported by: John Taylor
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Well, thats embarrasing. I forgot to initialize the caller_id storage.
(closes issue ASTERISK-19311)
Reported by: tootai
Tested by: rmudgett
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Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651
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(closes issue ASTERISK-17053)
Reported by: justdave
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Review: https://reviewboard.asterisk.org/r/1707/
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Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as
a result.
Review: https://reviewboard.asterisk.org/r/1697/
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Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
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monitor launch.
MixMonitor receives a new option i(channel_variable) which stores the unique id at said
variable. StopMixMonitor now accepts ID as an optional argument, which if included will
make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI
commands and AMI actions have been ammended to work with the IDs as well. In addition,
monitors across a channel can now be listed be listed via CLI command "mixmonitor list
<channel>" which will display all of the mixmonitors active on that channel along with
the files they each have open. Created by Sergio González Martín.
(closes issue ASTERISK-19096)
Reported by: Sergio González Martín
Review: https://reviewboard.asterisk.org/r/1643/
Review: https://reviewboard.asterisk.org/r/1682/
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Don't be alarmed. This only affected trunk, and it would have
required manager access to your system.
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Previously, realtime queues could be loaded without defining the queue member
table. This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage. Revision 342223 broke
this when it changed the return value for realtime_multientry to return NULL
when no results are returned. Previously, an empty ast_config object was
expected.
(closes issue ASTERISK-19170)
Reported by: Rene Mendoza
Tested by: Rene Mendoza
Patches:
rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283)
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This patch addresses two issues in ConfBridge and the channel bridge layer:
1. It fixes a race condition wherein the bridge channel could be hung up
2. It removes the deadlock avoidance from the bridging layer and makes the
bridge_pvt an ao2 ref counted object
Patch by David Vossel (mjordan was merely the commit monkey)
(issue ASTERISK-18988)
(closes issue ASTERISK-18885)
Reported by: Dmitry Melekhov
Tested by: Matt Jordan
Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628)
(closes issue ASTERISK-19100)
Reported by: Matt Jordan
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1654/
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accepting endpoint is bridged.
Like Dial and Queue, FollowMe needs to deal with
AST_CONTROL_CONNECTED_LINE information so when the parties are initially
bridged, the connected line information will be correct.
* Added the 'I' option just like the app_dial and app_queue 'I' option.
* Made 'N' option ignored if the call is already answered.
(closes issue ASTERISK-18969)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1656/
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There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
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If SLA was reloaded without the config file being changed, current settings got
wiped out before the SLA reload code decided it wasn't going to reload the file
since nothing was changed. Moving the settings reset later in the reload
process fixes this.
(closes issue AST-744)
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Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 350024 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue ASTERISK-19055)
Reported by: Matt Jordan
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Merged revisions 349872 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 349873 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When looking up a ConfBridge profile, the config parser would, if it
found a channel datastore on the channel requesting the bridge profile,
unlock the channel mutex twice. Since that's a little aggressive,
it now only unlocks it once.
(closes issue ASTERISK-19042)
Reported by: Matt Jordan
Tested by: Matt Jordan
Patches:
19042 uploaded by David Vossel (license 5628)
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Merged revisions 349619 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user. It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.
(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1614/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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