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r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
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r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
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Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
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Add app_skel.c as an example in app.c and fix some formating for the "Dial Privacy scripts" so it actually shows up in the Doxygen output.
(issue ASTERISK-20259)
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The v1.8 -r369258 change to fix the F and F(x) action logic introduced a
regression in passing the hangup cause from the called channel to the
caller channel.
(closes issue ASTERISK-20287)
Reported by: Konstantin Suvorov
Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified)
Tested by: rmudgett
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This fixes three main issues
* Change asprintf() uses to ast_asprintf() so that it
pairs properly with ast_free() and no longer causes
MALLOC_DEBUG to freak out.
* When ast_asprintf() fails, set the pointer NULL if
it will be referenced later.
* Fix some memory leaks that were spotted while taking
care of the first two points.
(Closes issue ASTERISK-20135)
reported by Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2071
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The paragraph describing the SubEvent belongs with the SubEvent parameter
itself, and not with its enum values. The order of parsing was placing
the description after the last enum, which isn't correct.
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Hangup handlers are an alternative to the h extension. They can be used
in addition to the h extension. The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up. Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel. You
can attach multiple handlers that will execute in the order of most
recently added first.
(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2002/
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This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules. Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.
The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation. Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event. The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files. It generates
the final core-[lang].xml file.
As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.
Review: https://reviewboard.asterisk.org/r/1967/
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* Make non-normal dialplan execution routines be able to run on a hung up
channel. This is preparation work for hangup handler routines.
* Fixed ability to support relative non-normal dialplan execution
routines. (i.e., The context and exten are optional for the specified
dialplan location.) Predial routines are the only non-normal routines that
it makes sense to optionally omit the context and exten. Setting a hangup
handler also needs this ability.
* Fix Return application being able to restore a dialplan location
exactly. Channels without a PBX may not have context or exten set.
* Fixes non-normal execution routines like connected line interception and
predial leaving the dialplan execution stack unbalanced. Errors like
missing Return statements, popping too many stack frames using StackPop,
or an application returning non-zero could leave the dialplan stack
unbalanced.
* Fixed the AGI gosub application so it cleans up the dialplan execution
stack and handles the autoloop priority increments correctly.
* Eliminated the need for the gosub_virtual_context return location.
Review: https://reviewboard.asterisk.org/r/1984/
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This was essentially duplicated functionality where normal channels used
AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage
into AST_CAUSE_ANSWERED_ELSEWHER usage.
Review: https://reviewboard.asterisk.org/r/1944
(closes issue ASTERISK-19865)
Patch-by: Birger Harzenetter
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The Dial and Queue I option is intended to block connected line updates
and redirecting updates. However, it is a feature that when a call is
locally redirected, the I option is disabled if the redirected call runs
as a local channel so the administrator can have an opportunity to setup
new connected line information. Unfortunately, the Dial and Queue I
option is disabled for *all* forked calls if one of those calls is
redirected.
* Make the Dial and Queue I option apply to each outgoing call leg
independently. Now if one outgoing call leg is locally redirected, the
other outgoing calls are not affected.
* Made Dial not pass any redirecting updates when forking calls.
Redirecting updates do not make sense for this scenario.
* Made Queue not pass any redirecting updates when using the ringall
strategy. Redirecting updates do not make sense for this scenario.
* Fixed deadlock potential with chan_local when Dial and Queue send
redirecting updates for a local redirect.
* Converted the Queue stillgoing flag to a boolean bitfield.
(closes issue ASTERISK-19511)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1920/
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This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.
This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.
Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)
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Before this patch, the predial routine executes on the ;1 channel of a
local channel pair. Executing predial on the ;1 channel of a local
channel pair is of limited utility. Any channel variables set by the
predial routine executing on the ;1 channel will not be available when the
local channel executes dialplan on the ;2 channel.
* Create ast_pre_call() and an associated pre_call() technology callback
to handle running the predial routine. If a channel technology does not
provide the callback, the predial routine is simply run on the channel.
Review: https://reviewboard.asterisk.org/r/1903/
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Thanks to Mark Murawski for the initial patch and feature definition.
(closes issue ASTERISK-19548)
Reported by: Mark Murawski
Review: https://reviewboard.asterisk.org/r/1878/
Review: https://reviewboard.asterisk.org/r/1229/
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* Simplify some code in app_dial and app_queue by calling
ast_app_exec_macro() and ast_app_exec_sub().
* Fix minor locking issue in app_dial for post-answer macro/gosub
MACRO/GOSUB_RESULT=GOTO: handling.
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The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel
lock was originally obtained is overkill where ast_channel_lock_both() was
inlined.
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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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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/
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The code may be just fine, but it had not received a "ship it!" on
review board yet.
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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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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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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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Review: https://reviewboard.asterisk.org/r/1786/
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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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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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Review: https://reviewboard.asterisk.org/r/1753/
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Review: https://reviewboard.asterisk.org/r/1733/
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Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651
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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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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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Add missing <variable></variable> tags in app_dial documentation.
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r339512 | rmudgett | 2011-10-05 12:01:46 -0500 (Wed, 05 Oct 2011) | 9 lines
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r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011) | 1 line
Fix Dial F option notes formatting.
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r339145 | lmadsen | 2011-10-03 14:55:15 -0500 (Mon, 03 Oct 2011) | 13 lines
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r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) | 6 lines
Make documentation for Dial() options 'F' and 'F()' more clear.
(Closes issue ASTERISK-18646)
Reported by: Physis Heckman
Tested by: Richard Mudgett
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r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines
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r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines
Fix deadlock when using dummy channels.
Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
ast_channel_unref(). Using ast_channel_release() needlessly grabs the
channel container lock and can cause a deadlock as a result.
* Analyzed use of ast_dummy_channel_alloc() and made use
ast_channel_unref() when done with the dummy channel. (Primary reason for
the reported deadlock.)
* Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
locks. Chan_local could not perform deadlock avoidance correctly.
(Potential deadlock exposed by this issue. Secondary reason for the
reported deadlock since the held lock was part of the deadlock chain.)
* Fixed some uses of ast_dummy_channel_alloc() not checking the returned
channel pointer for failure.
* Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected
by testing the bogus_chan value.
* Fixed needlessly clearing a 1024 char auto array when setting the first
char to zero is enough in manager.c:action_getvar().
(closes issue ASTERISK-18613)
Reported by: Thomas Arimont
Patches:
jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Thomas Arimont
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r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines
Merged revisions 337118 via svnmerge from
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r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines
Fix for incorrect voicemail duration in external notifications
This patch fixes an issue where the voicemail duration was being reported
with a duration significantly less than the actual sound file duration.
Voicemails that contained mostly silence were reporting the duration of
only the sound in the file, as opposed to the duration of the file with
the silence. This patch fixes this by having two durations reported in
the __ast_play_and_record family of functions - the sound_duration and the
actual duration of the file. The sound_duration, which is optional, now
reports the duration of the sound in the file, while the actual full duration
of the file is reported in the duration parameter. This allows the voicemail
applications to use the sound_duration for minimum duration checking, while
reporting the full duration to external parties if the voicemail is kept.
(issue ASTERISK-2234)
(closes issue ASTERISK-16981)
Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443
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r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines
Merged revisions 336658 via svnmerge from
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r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines
Made Dial d and H options no longer immediately auto-answer the calling leg.
The Dial d and H options break DTMF attended transfer atxferdropcall
option.
1) Party A calls party B.
2) Party B does a DTMF attended transfer to Party C.
If the dialplan uses the Dial d or H options to call Party C then the Dial
application answers the call immediately before initiating the call leg to
Party C. The premature answer causes the transfer code to not invoke the
atxferdropcall=no behavior for a blonde transfer since Party C has
"answered". The transfer code thinks that Party B has "consulted" with
Party C when Party B hangs up and completes the transfer to Party A.
Party A now hears ringback until Party C actually answers.
ASTERISK-13294 Dial d option.
ASTERISK-11067 Dial H option to disconnect before answer.
The referenced issues made Dial answer with the d and H options because
many SIP and ISDN phones cannot send DTMF before the call is connected.
* Made require the dialplan to control when or if the call needs to be
answered to use the Dial application d and H options. (The call is no
longer surprise answered when using the Dial d or H options.)
Review: https://reviewboard.asterisk.org/r/1381/
JIRA AST-623
JIRA AST-666
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r335346 | kmoore | 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) | 17 lines
Merged revisions 335341 via svnmerge from
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r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines
Ensure frames are not written to dialed channel if ringback is requested
When a single channel was dialed and there was media to be forwarded to the
calling channel, the media was written without regard for ringback causing
silence to be heard in some circumstances. This regression was introduced
when the meaning of "single" changed to mean only the number of channels
dialed.
(closes issue ASTERISK-18083)
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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r331644 | jrose | 2011-08-12 11:18:57 -0500 (Fri, 12 Aug 2011) | 9 lines
Merged revisions 331635 via svnmerge from
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r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line
Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme
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r331579 | qwell | 2011-08-11 16:54:54 -0500 (Thu, 11 Aug 2011) | 13 lines
Merged revisions 331578 via svnmerge from
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r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines
Use proper values for 64-bit option flags.
Also, reusing bits es no bueno, so change the value of a duplicate.
(issue ASTERISK-18239)
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r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328663 via svnmerge from
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r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines
app_dial may double free a channel datastore
When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it.
(closes issue ASTERISK-17917)
Reported by: Mark Murawski
Tested by: Mark Murawski
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines
don't do native/remote bridging if a framehook is active on the channel
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