When a change in time occurs, such that the timestamps associated with frames
being placed into an adaptive jitter buffer (implemented in jitterbuf.c)
are significantly different then the previously inserted frames, the jitter
buffer checks to see if it needs to be resynched to the new time frame. If
three consecutive packets break the threshold, the jitter buffer resynchs
itself to the new timestamps. This currently only occurs when history is
calculated, and hence only on JB_TYPE_VOICE frames.
JB_TYPE_CONTROL frames, on the other hand, are never passed to the history
calculations. Because of this, if the jump in time is greater then the
maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are
dropped and no resynchronization occurs. Alterntively, if the overfill
logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the
buffer, but with a time reference that is not applicable. Subsequent
JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads
from the jitter buffer reach the errant JB_TYPE_CONTROL frame.
This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As
JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the
resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch
threshold.
Note that this only impacts chan_iax2, as other consumers of the adaptive
jitter buffer use the abstract jitter buffer API, which does not use
JB_TYPE_CONTROL frames.
Review: https://reviewboard.asterisk.org/r/1814/
(closes issue ASTERISK-18964)
Reported by: Kris Shaw
Tested by: Kris Shaw, Matt Jordan
Patches:
jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact
that logger.c implements 32 log levels (because of the custom log level stuff).
asterisk.c uses this define to size an array of levels per remote console.
This array is modified in ast_console_toggle_loglevel(), which is called by the
"logger set level" CLI command. While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to toggle a
custom log level on a remote console, as well. However, doing so led to an
invalid array index in asterisk.c.
This array is read from any time a log message is written to a console. So,
all custom log level messages resulted in a bogus read if a remote console
was connected.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The set_format() function was more subtle in how it modified the
struct ast_channel readtrans/writetrans values.
* Fixed ast_activate_generator() conversion correctly.
(closes issue ASTERISK-19434)
Reported by: Birger Harzenetter
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The process_output function in manager.c attempted to call fclose and close immediately
afterwards. Since fclose implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error checking to fclose and
close depending on which was deemed necessary. Also error messages. Thanks to Rosen
Iliev for pointing out the location of the problem.
(closes issue ASTERISK-18453)
Reported By: Jaco Kroon
Review: https://reviewboard.asterisk.org/r/1793/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive. This should also preserve the original case of
the device string as passed in to the event system. CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.
The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.
This adds a unit test to verify that the event system works as expected.
(closes issue ASTERISK-19422)
Review: https://reviewboard.asterisk.org/r/1780/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The check if an ISDN call is bridged before it could be placed on hold is
not necessary and is overly restrictive. The check was originally done to
prevent problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application. The ISDN transfer code has not required this restriction for
quite some time because ECT could transfer any two active calls to each
other.
* Remove ISDN hold restriction for calls connected to applications.
* Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
(closes issue ASTERISK-19388)
Reported by: Birger Harzenetter
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This takes two actions.
1. Move the reading of the alertpipe in __ast_read() to immediately before the
removal of frames from the readq. This means we won't do something silly like
read from the alertpipe, then ignore the fact that there's a frame to get from
the readq since channel's fdno is the AST_TIMING_FD.
2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the
channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because
if the rate is 0 and the timingfunc is NULL, it means that the channel's timing
fd is being invalidated, so any pending reads should not occur.
This may actually solve more issues than the referenced one below, but it's not
known at this time for sure.
(closes issue ASTERISK-19223)
reported by Frank-Michael Wittig
Review: https://reviewboard.asterisk.org/r/1779
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the
behavior of ast_find_ourip such that port number was wiped out. This caused
the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be
0. This change causes ast_find_ourip to be port-preserving again.
(closes issue ASTERISK-19430)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add the ability to specify what kind of locking an ao2 object has when it
is allocated. The locking could be one of: MUTEX, RWLOCK, or none.
New API:
ao2_t_alloc_options()
ao2_alloc_options()
ao2_t_container_alloc_options()
ao2_container_alloc_options()
ao2_rdlock()
ao2_wrlock()
ao2_tryrdlock()
ao2_trywrlock()
The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning
change. They no longer mean that the object is protected by an external
mechanism. They mean the lock associated with the object has already been
manually obtained by one of the ao2_lock calls. This change is necessary
for RWLOCK support since they are not reentrant. Also an operation on an
ao2 container may require promoting a read lock to a write lock by
releasing the already held read lock to re-acquire as a write lock.
Replaced API calls:
ao2_t_link_nolock()
ao2_link_nolock()
ao2_t_unlink_nolock()
ao2_unlink_nolock()
with the respective
ao2_t_link_flags()
ao2_link_flags()
ao2_t_unlink_flags()
ao2_unlink_flags()
API calls to be more flexible and to allow an anticipated enhancement to
control linking duplicate objects into a container.
The changes to format.c and format_cap.c are taking advantange of the new
ao2 locking options to simplify the use of the format capabilities
containers.
Review: https://reviewboard.asterisk.org/r/1554/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Occasionally there is a need to put all objects in one container also into
another container.
Some reasons you might need to do this:
1) You need to reconfigure a container. You would do this by creating a
new container with the new configuration and ao2_container_dup the old
container into it. Then replace the old container with the new. Then
destroy the old container.
2) You need the contents of a container to remain stable while operating
on all of the objects. You would do this by creating a cloned container
of the original with ao2_container_clone. The cloned container is a
snapshot of the objects at the time of the cloning. When done, just
destroy the cloned container.
Review: https://reviewboard.asterisk.org/r/1746/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix opaquification conversion error.
(closes issue ASTERISK-19424)
Reported by: Jeremy Pepper
Patches:
asterisk-19424-initialize_priority_regression.diff (license #5026) patch uploaded by Michael L. Young
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
chan_iax2 to pass in the correct types.
chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk
at this point, so this feels like a safe change to make.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp. This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.
This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library. From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately. This
was changed to account for the differences in handling remote and local
policies in libsrtp.
Review: https://reviewboard.asterisk.org/r/1741/
(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
(with some small modifications for this check-in)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Custom parking extensions may not be coded such that the first and only
extension priority is the Park application. These custom parking
extensions will not be recognized as parking extensions. When a call is
blind transferred to an extension that is not recognized as a parking
extension, the normal blind transfer code causes the transferred channel
to start executing dialplan. Calls that get parked in this manner do not
know the original channel name that parked the call so the original parker
could never be called back if the parked call is not retrieved before the
timeout time. The parking space is also announced to the call being
parked as a side effect of not knowing the original parking channel.
* Fix handling of BLINDTRANSFER channel variable for call parking.
* Fixed SIP blind transfer using the wrong dialplan context variable to
check for the parking extension.
(closes issue ASTERISK-19322)
Reported by: aragon
Tested by: rmudgett, jparker
Review: https://reviewboard.asterisk.org/r/1730/
JIRA AST-766
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If the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run, Asterisk
would crash.
This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to unload
them all via "module unload -h res_calendar.so".
Review: https://reviewboard.asterisk.org/r/1752/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The accessors names for the "emulate_dtmf_digit" field on the ast_channel
are misleading. Change them to ast_channel_dtmf_digit_to_emulate*.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is. The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.
Review: https://reviewboard.asterisk.org/r/1599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
