This patch adds a note to any parameter that has 'allowMultiple' set in
the Swagger documentation.
(closes issue ASTERISK-22704)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ring operation sends ringing to the specified channel it is invoked on.
The dtmf operation can be used to send DTMF digits to the specified channel
of a specific length with a wait time in between. Finally hangup reasons
allow you to specify why a channel is being hung up (busy, congestion).
Early media behavior has also been tweaked slightly. When playing media to a channel
it will no longer automatically answer. If it has not been answered a progress indication
is sent instead.
(closes issue ASTERISK-22701)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2916/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a Stasis application is specified an implicit subscription is done on the originated
channel. This was previously done with the channel lock held which is dangerous as the
underlying code locks the container and iterates items. This change releases the lock
on the originated channel before subscribing occurs.
(closes issue ASTERISK-22768)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2979/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a command to the command queue to explicitly depart the channel
from the bridge when it is told it has left. If the channel has already been departed
or has entered a different bridge this command will become a no-op.
(closes issue ASTERISK-22703)
Reported by: John Bigelow
(closes issue ASTERISK-22634)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/2965/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For awhile now, we've noticed continuous integration builds hanging on CentOS 6
64-bit build agents. After resolving a number of problems with symbols, strange
locks, and other shenanigans, the problem has persisted. In all cases, gdb
shows the Asterisk process stuck in loader.c on one of the infinite while loops
that calls dlclose repeatedly until success.
The documentation of dlclose states that it returns 0 on success; any other
value on error. It does not state that repeatedly calling it will eventually
clear those errors. Most likely, the repeated calls to dlclose was to force a
close by exhausting the references on the library; however, that will never
succeed if:
(a) There is some fundamental error at work in the loaded library that
precludes unloading it
(b) Some other loaded module is referencing a symbol in the currently loaded
module
This results in Asterisk sitting forever.
Since we have matching pairs of dlopen/dlclose, this patch opts to only call
dlclose once, and log out as an ERROR if dlclose fails to return success. If
nothing else, this might help to determine why on the CentOS 6 64-bit build agent
things are not closing successfully.
Review: https://reviewboard.asterisk.org/r/2970
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Based on feedback from ipengineer in #asterisk, when the media indexer
cannot access a sound file on the system (or otherwise fails) Asterisk
displays a "Cannot frob file" error but fails to tell you why. This is
especially problematic as the media_indexer failing will rpevent Asterisk
from starting, as it is in the core.
We now display the errno error messages so folks can figure out what they've
done wrong.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I neglected to implement two of the endpoint subscription functions when
I did the work. Normally, you'll only hit that when you unsubscribe from
a specific endpoint.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:
* channel.c: When copying variables from a parent channel to a child channel,
specify the channels involved. Do not log anything for a variable that is not
inherited; the fact that it doesn't have an _ or __ already signifies that it
won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
to use these debug messages, and for each format that is registered (on
startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
For short tests in the Asterisk Test Suite, this should make finding the
actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
Often, description elements - which are not required - are not provided.
This debug message adds no additional value, as it is not indicative of an
error or helpful in debugging which element did not contain a 'blah' element
as a child. If an element is supposed to contain a child element, then that
XML tree should have failed validation in the first place.
Review: https://reviewboard.asterisk.org/r/2966/
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When a bridge transitions away from one tech to another, the tech going
away is provided a dummy bridge with no channels in it to tear down.
Currently this means that the teardown code exits prematurely and does
not tear anything down. This change tears down RTP bridging for the
channel provided in the leave bridge tech callback.
This also reverts the majority of r400403 since it is now redundant.
(closes issue ASTERISK-22628)
(closes issue ASTERISK-22676)
Reported by: John Bigelow
Reported by: Kevin Harwell
Tested by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2905/
Patches:
native_rtp_fix.diff uploaded by Kinsey Moore (License 6273)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When merging in the patch for ASTERISK-22728, the UPGRADE.txt file was changed
incorrectly. That change should have gone into ASTERISK-11.txt.
This commit is to fix that.
Also, another comment in the UPGRADE-11.txt was missing and this commit adds
that as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing. The display of
"N" used to mean NAT (i.e. yes). The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear. That was a great suggestion.
Therefore, this patch does the following:
* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.
* A column for the 'Comedia' setting has been added. It too will display the
setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.
* UPGRADE.txt has been updated to document this change.
(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
asterisk-forcerport-display-clarification_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2941
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Surrogate channels would pop up from time to time in dial message handling.
This would cause a WARNING message to appear, indicating that the Surrogate
channel had no CDR. This patch filters out those channels that have the
internal implementation flag set, such that the WARNING message isn't
displayed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Removed another silly use of RAII_VAR(). RAII_VAR() and SCOPED_LOCK()
are not silver bullets that allow you to turn off your brain.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In function ast_rtp_instance_early _bridge_make_compatible the
use of instance 0/1 as arguments doesn't clearly communicate a
direction that the copying of payloads from the source channel
to the destination channel will occur, making it more probable
to have the arguments to ast_rtp_codecs_payloads_copy() put in
the reverse order. This patch renames the arguments with _dst
and _src suffixes and corrects the copy direction.
(closes issue ASTERISK-21464)
Reported by: Kevin Stewart
Review: https://reviewboard.asterisk.org/r/2894/
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Test shows rtpmap:119 being copied per this change, but is not in sip invite
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Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.
(closes issue ASTERISK-22005)
Reported by: Torrey Searle
Patches:
optional_avpf_trunk.patch uploaded by tsearle (license 5334)
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After a series of upgrades over recent weeks, I've discovered that
test_json.c won't compile in dev mode any more for me.
One of gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate
tempnam. Which, in general, is a good thing. But for test code that just
needs a temporary file, it's just annoying.
This patch replaces usage of tempname with mkstemp, avoiding the
deprecation warning. It also removes the temporary files when the test
is complete, which apparently we weren't doing before (oops).
Review: https://reviewboard.asterisk.org/r/2957/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
slightly different than my proposal. Instead of putting an 'extends'
field on the subtype, the base type has a 'subTypes' field, which is a
list of the subTypes. Given that its a messaging model and not an
object model, kinda makes sense.
This patch changes the events.json api-doc, and the python translators
to take the new format into account.
Other changes that are in Swagger 1.2 were not adopted, since the spec
is still in flux, and could change before it's finalized.
A summary of changes to the Swagger-1.2 spec can be found at
https://github.com/wordnik/swagger-core/wiki/1.2-transition.
(closes issue ASTERISK-22440)
Review: https://reviewboard.asterisk.org/r/2909/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses
was removed by mistake. This patch restores this functionality and adds notes
about IPv6 addresses in the sample config.
(closes issue ASTERISK-22741)
Reported by: Joshua Colp
Tested by: Michael L. Young
Patches:
asterisk-22741-fix-binding-multiple-addr.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2945/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r400089, a patch was put in to correct erroneous RTCP statistic resets.
Unfortunately, ast_rtp_read can be called on an RTP instance that does not
have RTCP information. This patch prevents that crash by only resetting
the statistics if we do actually have an RTCP instance.
(issue AST-1174)
(closes issue ASTERISK-22667)
Reported by: John Bigelow
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The queue_log entry resulting from CLI "queue remove member" when
log_membername_as_agent is enabled is wrong. It always uses the interface
name instead of the member name in the queue_log entry.
* Get the queue member before removing it from the queue so the member
name is available for the queue_log entry.
(closes issue ASTERISK-21826)
Reported by: Oscar Esteve
Patches:
fix_membername.diff (license #6505) patch uploaded by Oscar Esteve
(modified to fix potential ref leak)
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