Core/PBX: Deadlock between dialplan execution and application unregistration.
Not easy to reproduce, but we have noticed deadlocks when unloading a module
while dialplan is handling a request.
The deadlock is between :
1) Dialplan execution: pbx_extension_helper() first taking conlock,
then pbx_findapp() [when called] asking for lock on apps list.
2) Application unregistration: ast_unregister_application() first taking lock
on apps list, then unreference_cached_app() [when called] asking for conlock.
As a protection, I suggest to modify ast_unregister_application(), so that it
anticipates the need of conlock, before taking the lock on apps list.
The side effect is a longer unavailability of conlock when unregistering an
application.
Jenkins2 [Fri, 16 Jun 2017 16:51:41 +0000 (11:51 -0500)]
Merge changes from topic 'sdp_api_adjustments'
* changes:
SDP: Set the remote c= line in RTP instance.
SDP: Add t= line in sdp_create_from_state()
stream: Ignore declined streams for some topology calls.
George Joseph [Tue, 13 Jun 2017 16:33:34 +0000 (10:33 -0600)]
res_ari: Add "module loaded" check to ari stubs
The recent change to make the use of LOAD_DECLINE more consistent
caused res_ari to unload itself before declining if the ari.conf
file wasn't found. The ari stubs though still tried to use the
configuration resulting in segfaults.
This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests
to see if res_ari is actually loaded and causes the stubs to also
decline if it isn't. The macro was then added to the mustache
template's "load_module" function.
ASTERISK-27026 #close Reported-by: Ronald Raikes
Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d
Richard Mudgett [Thu, 15 Jun 2017 17:33:22 +0000 (12:33 -0500)]
chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read.
The construction of the returned string assumed incorrectly that the
supplied buffer would always be initialized as an empty string. If it is
not an empty string we could overrun the supplied buffer by the length of
the non-empty buffer string plus one. It is also theoreticaly possible
for the supplied buffer to be overrun by a string terminator during a read
operation even if the supplied buffer is an empty string.
* Fix the assumption that the supplied buffer would already be an empty
string. The buffer is not guaranteed to contain an empty string by all
possible callers.
Joshua Colp [Thu, 15 Jun 2017 12:32:32 +0000 (12:32 +0000)]
channel: Fix reference counting in ast_channel_suppress.
The ast_channel_suppress function wrongly decremented the
reference count of the underlying structure used to keep
track of what should be suppressed on a channel if the
function was called multiple times on the same channel.
This change cleans up the reference counting a bit so
this no longer occurs.
George Joseph [Wed, 14 Jun 2017 13:54:45 +0000 (07:54 -0600)]
res_rtp_asterisk: Fix ssrc change for rtcp srtp
It looks like there was a copy/paste error in ast_rtp_change_source
where if there was a rtcp srtp instance, instead of updating its
ssrc we were updating the srtp instance ssrc twice.
ASTERISK-27022 #close Reported-by: Michael Walton
Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095
Joshua Colp [Thu, 8 Jun 2017 19:38:51 +0000 (19:38 +0000)]
bridge: Add a deferred queue.
This change adds a deferred queue to bridging. If a bridge
technology determines that a frame can not be written and
should be deferred it can indicate back to bridging to do so.
Bridging will then requeue any deferred frames upon a new
channel joining the bridge.
This change has been leveraged for T.38 request negotiate
control frames. Without the deferred queue there is a race
condition between the bridge receiving the T.38 request
negotiate and the second channel joining and being in the
bridge. If the channel is not yet in the bridge then the T.38
negotiation fails.
A unit test has also been added that confirms that a T.38
request negotiate control frame is deferred when no other
channel is in the bridge and that it is requeued when a new
channel joins the bridge.
Kevin Harwell [Tue, 13 Jun 2017 19:17:29 +0000 (14:17 -0500)]
res_pjsip_refer/session: Calls dropped during transfer
When doing an attended transfer it's possible for the transferer, after
receiving an accepted response from Asterisk, to send a BYE to Asterisk,
which can then be processed before Asterisk has time to start and/or
complete the transfer process. This of course causes the transfer to not
complete successfully, thus dropping the call.
This patch makes it so any BYEs received from the transferer, after the REFER,
that initiate a session end are deferred until the transfer is complete. This
allows the channel that would have otherwise been hung up by Asterisk to
remain available throughout the transfer process.
Alexei Gradinari [Mon, 12 Jun 2017 14:57:24 +0000 (10:57 -0400)]
res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled
If sending unsolicited mwi to all endpoints on startup is disabled
(mwi_disable_initial_unsolicited=yes) do not need to create subscriptions.
If there are many (thousands) realtime endpoints configured with unsolicited mwi
and Vociemail Storage configured as ODBC or IMAP there will be huge number of
DB/IMAP requests on startup.
Guido Falsi [Thu, 8 Jun 2017 15:54:46 +0000 (17:54 +0200)]
BuildSystem: Add patches to allow building with recent LibreSSL
Add some #if defined checks which allow building against LibreSSL.
These patchess come from OpenBSD ports:
https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/
Joshua Colp [Wed, 7 Jun 2017 20:19:05 +0000 (20:19 +0000)]
chan_pjsip: Update device state when in early media.
The chan_pjsip module uses a calculation approach for
determining device state. This means that in situations
where we would expect device state to change we need to
tell the core to query. A scenario that was missed is
when early media was signaled.
This change adds the notification for the core to
query device state when we are told that early media
is being provided.
Joshua Colp [Tue, 6 Jun 2017 12:04:21 +0000 (12:04 +0000)]
pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.
PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.
This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.
Sean Bright [Tue, 6 Jun 2017 15:04:44 +0000 (11:04 -0400)]
res_rtp_multicast: Use consistent timestamps when possible
When a frame destined for a MulticastRTP channel does not have timing
information (such as when an 'originate' is done), we generate the RTP
timestamps ourselves without regard to the number of samples we are
about to send.
Instead, use the same method as res_rtp_asterisk and 'predict' a
timestamp given the number of samples. If the difference between the
timestamp that we generate and the one we predict is within a specific
threshold, use the predicted timestamp so that we end up with timestamps
that are consistent with the number of samples we are actually sending.
Joshua Colp [Wed, 31 May 2017 15:41:45 +0000 (15:41 +0000)]
res_pjsip: Add support for returning only reachable contacts and use it.
This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.
Kevin Harwell [Mon, 5 Jun 2017 16:27:32 +0000 (11:27 -0500)]
channel: ast_write frame wrongly freed after call to audiohooks
ASTERISK-26419 introduced a bug when calling ast_audiohook_write_list in
ast_write. It would free the frame given to ast_write if the frame returned
by ast_audiohook_write_list was different than the given one. The frame give
to ast_write should never be freed within that function. It is the caller's
resposibility to free the frame after writing (or when it its done with it).
By freeing it within ast_write this of course led to some memory corruption
problems.
This patch makes it so the frame given to ast_write is no longer freed within
the function. The frame returned by ast_audiohook_write_list is now subsequently
used in ast_write and is freed later. It is freed either after translate if the
frame returned by translate is different, or near the end of ast_write prior to
function exit.
Sean Bright [Wed, 31 May 2017 16:45:45 +0000 (12:45 -0400)]
pbx_builtin: Properly handle hangup during Background
Before this patch, when a user hung up during a Background, we would
stuff 0xff into a char and attempt a dialplan lookup of it. This caused
problems for some realtime engines which interpreted the value as the
beginning of an invalid UTF-8 sequence.
Sean Bright [Sun, 28 May 2017 20:43:12 +0000 (16:43 -0400)]
format: Reintroduce smoother flags
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Mark Michelson [Wed, 24 May 2017 15:09:22 +0000 (10:09 -0500)]
Confbridge: Add "sfu" video mode to bridge profile options.
A previous commit added plumbing to bridge_softmix to allow for an SFU
experience with Asterisk. This commit adds an option to app_confbridge
that allows for a confbridge to actually make use of the SFU video mode.
SFU mode is implemented in a "set it and forget it" kind of way. That
is, when the bridge is created, if SFU mode is enabled, then the video
mode gets set to SFU and cannot be changed. Future improvements may
allow for a hybrid experience (e.g. forward multiple video streams,
specifically those of the most recent talkers), but for this addition,
no such capability is present.
Mark Michelson [Fri, 5 May 2017 16:56:34 +0000 (11:56 -0500)]
Add primitive SFU support to bridge_softmix.
This sets up the "plumbing" in bridge_softmix to
be able to accommodate Asterisk asking as an SFU
(selective forwarding unit) for conferences.
The way this works is that whenever a channel enters or leaves a
conference, all participants in the bridge get sent a stream topology
change request. The topologies consist of the channels' original
topology, along with video destination streams corresponding to each
participants' source video streams. So for instance, if Alice, Bob, and
Carol are in the conference, and each supplies one video stream, then
the topologies for each would look like so:
Alice:
Audio,
Source video(Alice),
Destination Video(Bob),
Destination video (Carol)
Bob:
Audio,
Source video(Bob)
Destination Video(Alice),
Destination video (Carol)
Carol:
Audio,
Source video(Carol)
Destination Video(Alice),
Destination video (Bob)
This way, video that arrives from a source video stream can then be
copied out to the destination video streams on the other participants'
channels.
Once the bridge gets told that a topology on a channel has changed, the
bridge constructs a map in order to get the video frames routed to the
proper destination streams. This is done using the bridge channel's
stream_map.
This change is bare-bones with regards to SFU support. Some key features
are missing at this point:
* Stream limits. This commit makes no effort to limit the number of
streams on a specific channel. This means that if there were 50 video
callers in a conference, bridge_softmix will happily send out topology
change requests to every channel in the bridge, requesting 50+
streams.
* Configuration. The plumbing has been added to bridge_softmix, but
there has been nothing added as of yet to app_confbridge to enable SFU
video mode.
* Testing. Some functions included here have unit tests.
However, the functionality as a whole has only been verified by
hand-tracing the code.
* Selectivenss. For a "selective" forwarding unit, this does not
currently have any means of being selective.
* Features. Presumably, someone might wish to only receive video from
specific sources. There are no external-facing functions at the moment
that allow for users to select who they receive video from.
* Efficiency. The current scheme treats all video streams as being
unidirectional. We could be re-using a source video stream as a
desetnation, too. But to simplify things on this first round, I did it
this way.
Sean Bright [Tue, 30 May 2017 14:34:34 +0000 (10:34 -0400)]
format_mp3: Re-work menuselect/build issues
Rather than removing format_mp3 from ALL_C_MODS (which caused format_mp3
to not show up in menuselect), use .PHONY targets when the necessary
source files are not present.
George Joseph [Tue, 30 May 2017 14:43:49 +0000 (08:43 -0600)]
test_json: Fix test names with reserved words
Some of the test names were actually reserved words (true, false,
int, null, string, bool). When the jenkins test results analyzer
does its thing it tries to create a map using the test names as
keys and fails because they're reserved words.
George Joseph [Wed, 24 May 2017 20:50:56 +0000 (14:50 -0600)]
unittests: Add a unit test that causes a SEGV and...
...that can only be run by explicitly calling it with
'test execute category /DO_NOT_RUN/ name RAISE_SEGV'
This allows us to more easily test CI and debugging tools that
should do certain things when asterisk coredumps.
To allow this a new member was added to the ast_test_info
structure named 'explicit_only'. If set by a test, the test
will be skipped during a 'test execute all' or
'test execute category ...'.
Sean Bright [Tue, 23 May 2017 20:42:04 +0000 (16:42 -0400)]
res_agi: Allow configuration of audio format of EAGI pipe
This change allows the format of the EAGI audio pipe to be changed by
setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of
the loaded formats.
Sean Bright [Tue, 23 May 2017 18:06:22 +0000 (14:06 -0400)]
res_agi: Prevent crash when SET VARIABLE called without arguments
Explicitly check that the appropriate number of arguments were passed to
SET VARIABLE before attempting to reference them. Also initialize the
arguments array to zeroes before populating it.
Joshua Colp [Mon, 15 May 2017 20:03:36 +0000 (20:03 +0000)]
app_queue: Fix members showing as being in call when not.
A change was done which added an 'in_call' flag to queue
members that was set to true while talking to an agent.
Unfortunately in practice this does not accurately reflect
whether they are talking to an agent or not. If a Local
channel is involved and a transfer is performed then the
app_queue application would incorrectly think the agent
was still in a call with the caller. This was done to
fix a race condition between an agent becoming available
by device state and the checking of the last call information
for the wrapup time. There was a small window where the
last call information would be the previous value instead
of the new one.
This change goes about fixing the original issue in a
different way by considering the call completed if device
state is received which would make the agent available
and if they are currently in a call. If this occurs the
last call information is updated before the agent becomes
available ensuring that old information is not present
when checking if the member should be called. This also
improves the transfer situation by actually updating
and enforcing the wrapup time.
Robert Mordec [Tue, 23 May 2017 10:45:29 +0000 (12:45 +0200)]
app_confbridge: Race between removing and playing name recording while leaving
When user leaves a conference, its channel calls async_play_sound_file()
in order to play the name announcement and then unlinks the sound file.
The async_play_sound_file() function adds a task to conference playback queue,
which then runs playback_common() function in a different thread.
It leads to a race condition when, in some cases, channel thread may unlink
the sound file before playback_common() had a chance to open it.
This patch creates a file deletion task, that is queued after playback.
Kevin Harwell [Mon, 22 May 2017 18:51:40 +0000 (13:51 -0500)]
res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm
When using rtcp mux if an rtcp payload came in it would still use the srtp
unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp
data was being passed to the rtp unprotect method this would result in an
error.
This patch ensures that the correct unprotect method is chosen by making
sure the passed in rtcp flag is appropriately set when rtcp mux is enabled
and an rtcp payload is received.
Sean Bright [Fri, 19 May 2017 15:05:36 +0000 (15:05 +0000)]
chan_sip: Better ICE handling for RTCP-MUX
If we are offered or are offering RTCP-MUX, don't consider RTCP ICE
candidates. This confuses certain browsers (current Firefox for
example) and causes intial audio setup delays.
Steve Davies [Fri, 12 May 2017 15:38:27 +0000 (16:38 +0100)]
app_queue: Add QUEUE_RAISE_PENALTY feature
Additional variable to work alongside QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY,
including an extra parameter in queuerules.conf. This value causes lower
Agent penalty values to "raise up" so that they can join higher penalty agents
and be treated equally after a period of time.
projects / thirdparty / asterisk.git / log
