Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
rtp instance destroy for chan_unistim. Also several fixes
for displayed text translation.
Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc
Correct typo of end-pints to end-points
Re-wrap session timer parameter docs to max 80 chars wide; this
eases reading on terminals with lower resolution, commonly the case
for those with visual impairments.
ASTERISK-26573
Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b
Signed-off-by: C.J. Collier <cjcollier@linuxfoundation.org>
res_pjsip_sesssion was hooking into transaction and invite state
changes. One of the reasons for doing so was due to the
PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the
message sending process, and so we should call session supplements to
alter the outgoing message.
In reality, this event was meant to indicate that the message either
a) had already been sent, or
b) required a DNS lookup and would be sent when the DNS query
completed.
In case (a), this meant we were altering an already-sent
request/response for no reason. In case (b), this potentially meant we
could be trying to alter a request/response at the same time that the
DNS resolution completed. In this case, it meant we might be stomping on
memory being used by the thread actually sending the message. This
caused potential crashes and memory corruption.
This patch removes the calls to session supplements from the case where
the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to
alter the message at this point is too late, and it can cause nothing
but harm to try to do it. Because there were no longer any calls to the
handle_outgoing() function, it has been removed.
Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92
This is another case where manual frame deferral can be replaced with
centralized routines instead.
Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e
(cherry picked from commit d149c4b9e0)
Rather than use manual frame deferral, just let the channel API do it
for us.
ASTERISK-26343
Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49
(cherry picked from commit 8ba3e2fc27)
AGI recently was modified to defer important frames. This was because
when AGI was used in a connected line interception routine, the
resulting connected line frame would end up getting discarded by the
AGI.
However, this caused bad behavior in other cases. Specifically, during a
transfer, if someone attempted to manually set the Caller ID on a
channel in an AGI, the deferred connected line frame would end up
overwriting what had been manually set in the AGI.
Since the initial issue was specific to interception routines, this
change removes the manual frame deferral from AGI and instead uses the
new frame deferral API in interception routines.
ASTERISK-26343 #close
Reported by Morton Tryfoss
Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208
There are several places in Asterisk that have duplicated logic
for deferring important frames until later.
This commit adds a couple of API calls to facilitate this automatically.
ast_channel_start_defer_frames(): Future reads of deferrable frames on
this channel will be deferred until later.
ast_channel_stop_defer_frames(): Any frames that have been deferred get
requeued onto the channel.
ASTERISK-26343
Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641
A NULL bridge has special meaning in res_stasis for
unsubscribing. It means that a subscription to ALL
bridges should be removed. This should not be done
as part of the normal subscription management in
the res_stasis channel loop.
ASTERISK-26468
Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0
This is a regression over Asterisk 11, introduced by
2dc8a06006. Previously, recordings started via
the automon DTMF code would automatically be mixed together using sox because
app_monitor would be called with the m option. This commit restores this
behavior.
Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759
Not surprisingly, using Respoke (and possibly other systems) it is
possible to blow past the 16k limit for a WebSocket packet size. This
patch bumps it up to 32k, which, at least for Respoke, is sufficient.
For now.
Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that
matter), this patch adds a LOW_MEMORY directive that sets the buffer to
8k for systems who have asked for their reduced memory availability to
be considered.
Change-Id: Id235902537091b58608196844dc4b045e383cd2e
When a bridge is created via ARI (through res_stasis), no video source
mode is set by default. As a result, any endpoint sending video media
won't ever see any video reflected back to it.
This patch defaults a bridge to a 'follow the talker' video mode.
Further work can be done to add routes that allow for the video mode to
be controlled through the /bridges resource.
Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866
When a channel is made the video source, the bridge holds a reference to
it. Whenever the video source changes, that reference is released.
However, a ref leak does occur if the channel leaves the bridge (such as
being hung up) while it is the video source, as the bridge never
releases the ref in such a case.
This patch adds a line to the bridge_channel_internal_join routine such
that, when a channel finishes its time in the bridge, it notifies the
bridge via ast_bridge_remove_video_src that if it is a video source its
reference should be released.
ASTERISK-26555 #close
Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a
It's actually quite useful to see the source of a video stream change.
This doesn't happen terribly often, even with talk detection - but when
it does, it's nice to know which channel is now providing your video
stream.
As a verbose 5 level message, it shouldn't be terribly spammy or costly
to have, and is 'lower level' then most other verbose messages that the
bridge system emits.
ASTERISK-26555
Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c
WebRTC clients really, really want to know the SSRC of the media they're
getting. Changing the SSRC is generally not a good thing.
bridge_softmix, starting in Asterisk 12, started changing the SSRC of
parties as they joined or left the bridge. With most phones, this isn't
a problem: phones just play back the stream they're getting. With WebRTC
clients, however, the SSRC is tied to a media stream that may be
negotiated. When a new SSRC just shows up, the media can be dropped.
As it turns out, the SSRC change shouldn't even be necessary. From the
perspective of the client, it's still talking to Asterisk with the same
media stream: why indicate that the far party has suddenly changed to a
different source of media?
This patch opts to just remove the SSRC changes. With this patch, video
clients that join/leave a softmix bridge actually get the video stream
instead of freaking out.
ASTERISK-26555
Change-Id: I27fec098b32e7c8718b4b65f3fd5fa73527968bf
The readdir_r function has been deprecated and should no longer be used. This
patch removes the readdir_r dependency (replaced it with readdir) and also moves
the directory search code to a more centralized spot (file.c)
Also removed a strict dependency on the dirent structure's d_type field as it
is not portable. The code now checks to see if the value is available. If so,
it tries to use it, but defaults back to using the stats function if necessary.
Lastly, for most implementations of readdir it *should* be thread-safe to make
concurrent calls to it as long as different directory streams are specified.
glibc falls into this category. However, since it is possible that there exist
some implementations that are not safe, locking has been added for those other
than glibc.
ASTERISK-26412
ASTERISK-26509 #close
Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba
This reverts commit 93332cb1d0.
Unfortunately, the aforementioned commit caused a regression (incoming calls
would eventually disconnect). Thus it is being removed.
ASTERISK-26523 #close
ASTERISK-25270
Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d
Fix logic on read second part of H.225 packet. There was infinite loop on
wrong connections due to read before poll.
Change-Id: I42b4bf75c46e4a5c5df5c5ca1f0bd74b8944e7ff
libresample is only needed by pjproject if we're building pjsua, which
we only do if TEST_FRAMEWORK is selected. It's required by pjsua to
process audio which is needed by some testsuite tests. Unfortunately,
pjproject relies on a newer version of libresample than the version
that ships by most distros so we need to compile the version that's
bundled with pjproject. Since we only need it for pjsua, we DON'T want
it's symbols exposed when we actually build asterisk.
There was a problem however... TEST_FRAMEWORK is only known AFTER we've
already run ./configure on both asterisk and pjproject but pjproject's
./configure needs to test it to know whether to set up to build
libresample or not. The previous way of figuring this out was to
always tell ./configure "yes" but not actually build the library. This
caused an issue where building libasteriskpj was being told to include
libresample but it wasn't actually there.
The solution is to still do a default pjproject configure during an
asterisk ./configure but if makeopts or menuselect.makeopts changes
subsequently, we now reconfigure pjproject, taking into account the
current state of TEST_FRAMEWORK. Previously, if makeopts or
menuselect.makeopts changed, only a recompile of pjproject was done.
Change-Id: I9b5d84c61384a3ae07fe30e85c49698378cc4685
Igor Goncharovskiy
zuul
Richard Mudgett
C.J. Collier
Joshua Colp
George Joseph
Sebastian Gutierrez
Mark Michelson
Alexander Anikin
Michael Kuron
Matt Jordan
Kevin Harwell