Dynamic payload types were statically defined in Asterisk. This unfortunately
limited the number of dynamic payloads that could be registered. With this patch
dynamic payload type numbers are now assigned dynamically and per RTP instance.
However, in order to limit any issues where some clients expect the old
statically defined value this patch makes it so the value Asterisk used to pre-
designate is used for the dynamic assignment if available.
An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf)
that turns the new dynamic behavior on or off. When off it reverts back to using
statically defined payload values. This option defaults to "yes" in Asterisk 15.
ASTERISK-26515 #close
patches:
ASTERISK-26515.diff submitted by jcolp (license 5000
Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc
The CDR code previously did not allow the user field to be set
from the 'h' extension in the dialplan. This change removes that
limitation and allows it to be set.
ASTERISK-26818
Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6
We aren't validating that the URI we just parsed is a SIP/SIPS one before
trying to access the user, host, and port members of a possibly uninitialized
structure.
Also update the MessageSend documentation to indicate what 'from' formats are
accepted.
ASTERISK-26484 #close
Reported by: Vinod Dharashive
Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30
Fixed a bug in function "ast_audiohook_write_frame" that checked the
variable other_factory_samples and only flushed the factories, so they
would be in sync, when other_factory_samples > 0. When there is not any
rtp incoming the variable other_factory_samples will be 0, and although
the result of "our_factory_ms - other_factory_ms" may be very large,
this led to the record file not syncing.
ASTERISK-26875 #close
Reported-by: Aaron An
Tested-by: Aaron An
Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22
POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.
Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.
Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48
Dereferencing struct ast_autochan.chan without first calling
ast_autochan_channel_lock() is unsafe because the pointer could change at
any time due to a masquerade. Unfortunately, ast_autochan_channel_lock()
itself uses struct ast_autochan.chan unsafely and can result in a deadlock
if the original channel happens to get destroyed after a masquerade in
addition to the pointer getting changed.
The problem is more likely to happen with v11 and earlier because
masquerades are used to optimize out local channels on those versions.
However, it could still happen on newer versions if the channel is
executing a dialplan application when the channel is transferred or
redirected. In this situation a masquerade still must be used.
* Added a lock to struct ast_autochan to safely be able to use
ast_autochan.chan while trying to get the channel lock in
ast_autochan_channel_lock(). The locking order is the channel lock then
the autochan lock. Locking in the other direction requires deadlock
avoidance.
* Fix unsafe ast_autochan.chan usages in app_mixmonitor.c.
* Fix unsafe ast_autochan.chan usages in app_chanspy.c.
* app_chanspy.c: Removed unused autochan parameter from next_channel().
ASTERISK-26867
Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592
Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range. e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.
The buffer overwrite is fixed two ways in this patch.
1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens. Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set. Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.
2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.
ASTERISK-26668
Change-Id: I555d97411140e47e0522684062d174fbe32aa84a
* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.
Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48
This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.
Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:
"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."
ASTERISK-25237 #close
Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70
This change adds a few things to facilitate stream topology changing:
1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.
2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.
Tests have also been written which confirm the multistream and
non-multistream behavior.
ASTERISK-26839
Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9
* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension. Ran into this when
developing a testsuite test. The AMI event ExtensionStatus came out with
the hint header value containing garbage. The AMI event PresenceStatus
also had the same issue.
* manager.c:action_extensionstate() no need to completely initialize the
hint[]. Only initialize the first element.
* pbx.c:ast_add_hint() Remove unnecessary assignment.
* chan_sip.c: Eliminate an unneeded hint[] local variable. We only care
about the return value of ast_get_hint() there.
Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b
* Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely
on read_stream being set to indicate a multi stream channel.
* Added ast_channel_is_multistream convenience function.
* Fixed issue where stream and default_stream weren't being set on
a frame retrieved from the queue.
* Now testing for NULL being returned from the driver's read or
read_stream callback.
* Fixed issue where the dropnondefault code was crashing on a
NULL f.
* Now enforcing that if either read_stream or write_stream are
set when ast_channel_tech_set is called that BOTH are set.
* Added the unit tests.
ASTERISK-26816
Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2
This introduces and documents the various states in the state machine.
This also introduces API functions that induce state changes, and places
TODO comments telling what needs to be done in addition to what is
already there. Those TODOs will be replaced with real code in upcoming
changes.
Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed
In the event that a cache file is removed out from under us, we should
treat the cache entry as stale and force a refresh.
ASTERISK-26774 #close
Reported by: Igor Gamayunov
Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52
... and clean them both up on uninstall.
We've fixed the issue where 'make install' was installing to
/usr/lib on 64-bit systems that use /usr/lib64. Now we need
to clean up the remnants in /usr/lib.
* 'make install' now prints a warning if DESTDIR/ASTLIBDIR
contains 'lib64' and libasterisk* shared libraries or modules
are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed
to 'lib'.
* 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and
DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'.
ASTERISK-26705
Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f
This change introduces an ast_read_stream function and callback in
the channel technology which allows reading frames from all streams
and not just the default streams.
The stream number has also been added to frames. This is to allow the
case where frames are queued onto the channel instead of being read
directly from the driver.
This change does impose a restriction on reading though: a chain of
frames can only contain frames from the same stream.
ASTERISK-26816
Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce
On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory. To make matters worse, options were being passed
to ldconfig on both Linux and FreeBSD that actually prevented
the rebuild of the cache.
* Fedora has a /usr/share/config.site that automatically tells
autoconf to use /usr/lib64 but CentOS does not. This logic was
copied to configure.ac and modified so systems like Ubuntu,
which still use /usr/lib for 64-bit systems, aren't affected.
Now that we have them in the correct directory...
In order for the system loader to find libasteriskssl and
libasteriskpj, one of 3 things has to happen...
- The linker cache must be rebuilt including the directory
where the libasterisk* libraries were installed. Only root
can rebuild the cache. This was busted.
- We have to link the asterisk binary with an rpath pointing
to the directrory where the libasterisk* libraries were
installed. This makes things very complicated and will happen
over the collective dead bodies of everyone who's had to
package a distribution with an rpath.
- Finally, you can start asterisk with LD_LIBRARY_PATH set to the
directrory where the libasterisk* libraries were installed.
There are no other options. So...
* The invokation of ldconfig has been moved from main/Makefile
to ASTTOPDIR/Makefile, the options have been removed, and
DESTDIR/ASTLIBDIR appended. If you aren't root, you will be
warned after the "Asterisk Installation Compete" banner that
you must re-run 'make install' as root, manually run
'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with
LD_LIBRARY_PATH.
ASTERISK-26705
Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982
This change adds an ast_write_stream function which allows
writing a frame to a specific media stream. It also moves
ast_write() to using this underneath by writing media
frames provided to it to the default streams of the channel.
Existing functionality (such as audiohooks, framehooks, etc)
are limited to being applied to the default stream only.
Unit tests have also been added which test the behavior of
both non-multistream and multistream channels to confirm that
the write() and write_stream() callbacks are invoked
appropriately.
ASTERISK-26793
Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c
Adds binaural synthesis to bridge_softmix (via convolution using libfftw3).
Binaural synthesis is conducted at 48kHz.
For a conference, only one spatial representation is rendered.
The default rendering is applied for mono-capable channels.
ASTERISK-26292
Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf
The "core show channel" CLI command will now output the streams
present on the channel with their details.
ASTERISK-26811
Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a
This establishes the basic allocation/destruction of an SDP state
object, plus some of the simpler getter methods involved. Subsequent
tasks will deal with adding a state machine, creating SDPs from
capabilities and options, and merging SDPs into a joint SDP.
Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709
On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.
This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.
If DESTDIR is specified, however, the old logic is executed as
the install process may not have permission to alter the ldconfig
cache.
ASTERISK-26705
Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4
When AMI encounters an error at the beginning of a session, it would
explicitly call ast_iostream_close() on its tcptls session's iostream.
It then would jump to a label where it would shut down the tcptls
session instance. The tcptls session instance would again attempt to
close the iostream.
Under normal circumstances, this might go by unnoticed. However, when
MALLOC_DEBUG is enabled, all fields on the iostream get set to
0xdeaddead when the iostream is freed. Thus a second call to
ast_iostream_close() after the iostream has been freed would reslt in an
attempt to call SSL_shutdown on 0xdeaddead, which would crash and burn
horribly.
The fix here is to not directly close the iostream from the dangerous
scenarios. The specific scenarios are:
* Exceeding the configured authlimit
* Failing to build a mansession on a new connection
Change-Id: I908f98d516afd5a263bd36b072221008a4731acd
This creates the following:
* Asterisk's internal representation of an SDP
* An API for translating SDPs from one format to another
* An implementation of a translator for PJMEDIA
Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b
This is step one of adding an SDP API: defining some
configurable settings for SDPs. This is based on options
that are currently supported in Asterisk.
Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7
On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.
This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.
ASTERISK-26705
Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519
This change adds unit tests for the various API calls relating
to stream topologies. This includes creation, destruction,
inspection, and manipulation.
Through this a few bugs were uncovered in the implementation:
1. Creating a topology using a format capabilities would fail as
the code considered a return value of 0 from the append stream
function to indicate an error which is incorrect.
2. Not all functions which placed a stream into a topology
set the position on the stream itself.
3. Appending a stream would cause a frack if the position
provided was the last one. This occurred because the existing
stream was queried but the index was outside of what the
vector was currently at for size.
ASTERISK-26786
Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0
* app_minivm: Use built-in completion facilities to complete optional
arguments.
* app_voicemail: Use built-in completion facilities to complete
optional arguments.
* app_confbridge: Add missing colons after 'Usage' text.
* chan_alsa: Use built-in completion facilities to complete optional
arguments.
* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'
* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'
* func_odbc: Correct completions for 'odbc read' and 'odbc write'
* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.
* main/bridge: Correct completions for 'bridge kick.'
* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.
* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'
* main/pbx_app: Remove redundant completions for 'core show
applications.'
* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'
* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.
Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
This change adds the media stream topology definition and API for
accessing and using it.
Some refactoring of the stream was also done.
ASTERISK-26786
Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568
The ast_waitfor_nandfds operation will manipulate the flags
of channels passed in. This was previously done without
the channel lock being held. This could result in incorrect
values existing for the flags if another thread manipulated
the flags at the same time.
This change locks the channel during flag manipulation.
ASTERISK-26788
Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed
In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c49. This patch
restores that behavior.
This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.
ASTERISK-26115 #close
Reported by: Nasir Iqbal
Change-Id: I8afbd06725e99610e02adb529137d4800c05345d
We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.
* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging. Made hold the channel lock after the called
party answers while updating the caller channel staging.
* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.
* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.
* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.
Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a
This change adds the media stream definition and API for
accessing and using it. Unit tests have also been written
which exercise aspects of the API.
ASTERISK-26773
Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87
When performing an SRV lookup using the ast_srv_lookup function it
did not properly handle the situation where 0 records are returned.
If this happened it would wrongly assume that at least one record
was present.
This change fixes the code so it will exit early if an error occurs
or if 0 records are returned.
ASTERISK-26772
patches:
srv_lookup.patch submitted by nappsoft (license 6822)
Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351
In ari.conf, when setting the option channelvars, every Stasis channel
snapshot would create a list of variable/value that would not be freed
when the snapshot is freed, resulting in a often-recurring memory
leak.
ASTERISK-26767 #close
Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d
OpenSSL 1.1 requires no explicit initialization. The hacks in the
library are not needed. They also happen to fail running Asterisk.
Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100
OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous
version-specific methods (such as TLSv1_client_method(). Other than
being simpler to use and more correct (gain support for TLS newer that
TLS1, in our case), the older ones produce a deprecation warning that
fails the build in dev-mode.
Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07
Using the timerfd timing module can cause channel freezing, lingering, or
deadlock issues. The problem is because this is the only timing module
that uses an associated alert-pipe. When the alert-pipe becomes
unbalanced with respect to the number of frames in the read queue bad
things can happen. If the alert-pipe has fewer alerts queued than the
read queue then nothing might wake up the thread to handle received frames
from the channel driver. For local channels this is the only way to wake
up the thread to handle received frames. Being unbalanced in the other
direction is less of an issue as it will cause unnecessary reads into the
channel driver.
ASTERISK-26716 is an example of this deadlock which was indirectly fixed
by the change that found the need for this patch.
* In channel.c:__ast_queue_frame(): Adding frame lists to the read queue
did not add the same number of alerts to the alert-pipe. Correspondingly,
when there is an exceptionally long queue event, any removed frames did
not also remove the corresponding number of alerts from the alert-pipe.
ASTERISK-26632 #close
Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6
There are several issues with deferring frames that are caused by the
refactoring.
1) The code deferring frames mishandles adding a deferred frame to the
deferred queue. As a result the deferred queue can only be one frame
long.
2) Deferrable frames can come directly from the channel driver as well as
the read queue. These frames need to be added to the deferred queue.
3) Whoever is deferring frames is really only doing the __ast_read() to
collect deferred frames and doesn't care about the returned frames except
to detect a hangup event. When frame deferral is completed we must make
the normal frame processing see the hangup as a frame anyway. As such,
there is no need to have varying hangup frame deferral methods. We also
need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real.
That fake hangup is to cause the PBX thread to break out of loops to go
execute a new dialplan location.
4) To properly deal with deferrable frames from the channel driver as
pointed out by (2) above, means that it is possible to process a dialplan
interception routine while frames are deferred because of the
AST_CONTROL_READ_ACTION control frame. Deferring frames is not
implemented as a re-entrant operation so you could have the unsupported
case of two sections of code thinking they have control of the media
stream.
A worse problem is because of the bad implementation of the AMI PlayDTMF
action. It can cause two threads to be deferring frames on the same
channel at the same time. (ASTERISK_25940)
* Rather than fix all these problems simply revert the API refactoring as
there is going to be only autoservice and safe_sleep deferring frames
anyway.
ASTERISK-26343
ASTERISK-26716 #close
Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496
A dialplan intercept routine is equivalent to an interrupt routine. As
such, the routine must be done quickly and you do not have access to the
media stream. These restrictions are necessary because the media stream
is the responsibility of some other code and interfering with or delaying
that processing is bad. A possible future dialplan processing
architecture change may allow the interception routine to run in a
different thread from the main thread handling the media and remove the
execution time restriction.
* Made res_agi.c:run_agi() running an AGI in an interception routine run
in DeadAGI mode. No touchy channel frames.
ASTERISK-25951
ASTERISK-26343
ASTERISK-26716
Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43
If an audiohook is placed on a channel that does not require transcoding,
muting that hook will cause the underlying frames to be muted as well.
The original patch is from David Woolley but I have modified slightly.
ASTERISK-21094 #close
Reported by: David Woolley
Patches:
ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded
by David Woolley
Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed
The escalator works by creating a set of startup commands in cli.conf
that set up logger channels and issue the debug commands for the
subsystems specified. If asterisk is running when it is executed,
the same commands will be issued to the running instance. The original
cli.conf is saved before any changes are made and can be restored by
executing '$prog --reset'.
The log output will be stored in...
$astlogdir/message.$uniqueid
$astlogdir/debug.$uniqueid
$astlogdir/dtmf.$uniqueid
$astlogdir/fax.$uniqueid
$astlogdir/security.$uniqueid
$astlogdir/pjsip_history.$uniqueid
$astlogdir/sip_history.$uniqueid
Some minor tweaks were made to chan_sip, and res_pjsip_history
so their history output could be send to a log channel as packets
are captured.
A minor tweak was also made to manager so events are output to verbose
when "manager set debug on" is issued.
Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543
Issue introduced in b59956a87. In the non-darwin case libastssl/pj
should be versioned. This causes the symbol file for this lib
to not be generated.
Change-Id: Ib07ae8c40252813c488e2c1ac6204fd42816dd4c
(cherry picked from commit 54b027916a)
* channel.c:ast_sendtext(): Fix T.140 SendText memory leak.
* format_compatibility.c: T.140 RED and T.140 were swapped.
* res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak.
* res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic
scheduled red_write().
* res_rtp_asterisk.c: Some other minor misc tweaks.
Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb
* make_silence() created a malloced silence slin frame without adding a
slin format ref. When the frame is destroyed it will unref the slin
format that never had a ref added. Memory corruption is expected to
follow.
* Simplified and fixed counting the number of samples in a frame list for
make_silence().
* Eliminated an unnecessary RAII_VAR associated with the make_silence()
frame.
Change-Id: I47de3f9b92635b7f8b4d72309444d6c0aee6f747
* ast_frisolate() could leak frame format refs on allocation
failures.
* Similified code in ast_frisolate() and code used by
ast_frisolate().
Change-Id: I79566d4d36b3d7801bf0c8294fcd3e9a86a2ed6d
The mechanism used for detecting the maximum log level compiled into the
linked pjproject did not work. The API call simply stores the requested
level into an integer and does no range checking. Asterisk was assuming
that there was range checking and limited the new value to the allowable
range. To get the actual maximum log level compiled into the linked
pjproject we need to get and save off the initial set log level from
pjproject. This is the maximum log level supported.
* Get and save off the initial log level setting before altering it to the
desired level on startup. This has to be done by a macro rather than
calling a core function to avoid incorrectly linking pjproject.
* Split the initial log level warning messages to warn if the linked
pjproject cannot support the requested startup level and if it is too low
to get the pjproject buildopts for "pjproject show buildopts".
* Adjust the CLI "pjproject set log level" to check the saved max log
level and to generate normal output messages instead of a warning message.
ASTERISK-26743 #close
Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4
The 'ari set debug' command has been enhanced to accept 'all' as an
application name. This allows dumping of all apps even if an app
hasn't registered yet. To accomplish this, a new global_debug global
variable was added to res/stasis/app.c and new APIs were added to
set and query the value.
'ari set debug' now displays requests and responses as well as events.
This required refactoring the existing debug code.
* The implementation for 'ari set debug' was moved from stasis/cli.{c,h}
to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted.
* In order to print the body of incoming requests even if a request
failed, the consumption of the body was moved from the ari stubs
to ast_ari_callback in res_ari.c and the moustache templates were
then regenerated. The body is now passed to ast_ari_invoke and then
on to the handlers. This results in code savings since that template
was inserted multiple times into all the stubs.
An additional change was made to the ao2_str_container implementation
to add partial key searching and a sort function. The existing cli
code assumed it was already there when it wasn't so the tab completion
was never working.
Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf
(cherry picked from commit 1d890874f3)
This change adds experimental support for providing RTCP
feedback information to codec modules so they can dynamically
change themselves based on conditions.
ASTERISK-26584
Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857
It was possible for a frame to be re-inserted into a jitter buffer after it
had been removed from it. A case when this happened was if a frame was read
out of the jitterbuffer, passed to the translation core, and then multiple
frames were returned from said translation core. Upon multiple frames being
returned the first is passed on, but sebsequently "chained" frames are put
back into the read queue. Thus it was possible for a frame to go back into
the jitter buffer where this would cause problems.
This patch adds a flag to frames that are inserted into the channel's read
queue after translation. The abstract jitter buffer code then checks for this
flag and ignores any frames marked as such.
Change-Id: I276c44edc9dcff61e606242f71274265c7779587
The task processor queue reached X scheduled tasks message was originally
intended to get logged only once per task processor to prevent spamming
the log. This is no longer necessary since high and low water thresholds
can better control when the message is logged.
It is beneficial to generate the warning each time a task processor
reaches the high water level because PJSIP stops processing new requests
while any high water alert is active. Without this change you would have
to enable at least debug level 3 logging to know about a repeated alert
trigger.
* Made generate the warning message whenever a task is pushed into the
task processor that triggers the high water alert.
* Appended 'again' to the warning for a repeated high water alert trigger.
Change-Id: Iabf75a004f7edaf1e5e8c323099418e667cac999
Adds the ability for extensions to be registered to include filename and
line number so that dialplan show output can show the filename and line
number of a config file responsible for generating a given extension.
This only affects config modules that are written to use the new extension
registering functions. In this patch, that only includes pbx_config, so
extensions registered in extensions.conf and any included extension will
be shown in this manner. Extensions registered in this manner will show
the filename and line number *instead* of the registrar.
ASTERISK-26658 #close
Reported by: Jonathan R. Rose
Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30
This is a semi-regression caused by the iostreams change. Prior to
iostreams, HTTP headers were written to a FILE handle using fprintf.
Then the body was written using a call to fwrite(). Because of internal
buffering, the result was that the HTTP headers and body would be sent
out in a single write to the socket.
With the change to iostreams, the HTTP headers are written using
ast_iostream_printf(), which under the hood calls write(). The HTTP body
calls ast_iostream_write(), which also calls write() under the hood.
This results in two separate writes to the socket.
Most HTTP client libraries out there will handle this change just fine.
However, a few of our testsuite tests started failing because of the
change. As a result, in order to reduce frustration for users, this
change alters the HTTP code to write the headers and body in a single
write operation.
ASTERISK-26629 #close
Reported by Joshua Colp
Change-Id: Idc2d2fb3d9b3db14b8631a1e302244fa18b0e518
ast_iostream_printf() attempts first to use a fixed-size buffer to
perform its printf-like operation. If the fixed-size buffer is too
small, then a heap allocation is used instead. The heap allocation in
this case was exactly the length of the string to print. The issue here
is that the ensuing call to vsnprintf() will print a NULL byte in the
final space of the string. This meant that the final character was being
chopped off the string and replaced with a NULL byte. For HTTP in
particular, this caused problems because HTTP publishes the expected
Contact-Length. This meant HTTP was publishing a length one character
larger than what was actually present in the message.
This patch corrects the issue by adding one to the allocation length.
ASTERISK-26629
Reported by Joshua Colp
Change-Id: Ib3c5f41e96833d0415cf000656ac368168add639
Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to
the CFLAGS. Not sure how they went missing.
Also fixed an uninstall problem where we weren't removing the
symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was
there, I fixed it for libasteriskssl as well.
Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556
OpenSSL 1.1.0 includes some major changes in the interface. See
https://wiki.openssl.org/index.php/1.1_API_Changes .
Status: Right now there are still a few deprecation notes with OpenSSL
1.1.0. But it's a start.
Changes:
* CRYPTO_LOCK is no longer available. Replace it with its value for now.
I don't completely understand what it is used for there.
* Remove several functions from libasteriskssl that seem to no longer be
needed.
* Structures have become opaque and are accesses with accessors.
* ERR_remove_thread_state() no longer needed.
* SSLv2 code now could no longer be used in 1.1.
ASTERISK-26109 #close
Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b
The latest Release candidate fails to create RTP streams when IPv6
is not available. Due to the changes made in September the ast_sockaddr
structure passed around to create these streams is always of AF_INET6
type, causing failure when used for IPv4. This patch adds a utility
function to check for availability of IPv6 and applies such check
at startup to determine how to create the ast_sockaddr structures.
ASTERISK-26617 #close
Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e
Use of the new logging is as simple as issuing the new CLI command or
setting the new pjproject.conf option.
Other options that can affect the logging are how you have the pjproject
log levels mapped to Asterisk log types in pjproject.conf and if you have
configured Asterisk to log the DEBUG type messages. Altering the
pjproject.conf level mapping shouldn't be necessary for most installations
as the default mapping is sensible. Configuring Asterisk to log the DEBUG
message type is standard practice for collecting debug information.
* Added CLI "pjproject set log level" command to dynamically adjust the
maximum pjproject log message level.
* Added CLI "pjproject show log level" command to see the currently set
maximum pjproject log message level.
* Added pjproject.conf startup section "log_level" option to set the
initial maximum pjproject log message level so all messages could be
captured from initialization.
* Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into
bundled pjproject. Pjproject will use the currently set run time log
level to determine if a log message is generated just like Asterisk
verbose and debug logging levels.
* In log_forwarder(), made always log enabled and mapped pjproject log
messages. DEBUG mapped log messages are no longer gated by the current
Asterisk debug logging level.
* Removed RAII_VAR() from res_pjproject.c:get_log_level().
ASTERISK-26630 #close
Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
The recent change that made frame deferral into an API had a behavior
change to it. When frame deferral was completed, we would take all of
the deferred frames and queue them all onto the channel in one call to
ast_queue_frame_head(). Before frame deferral was API-ized, places that
performed manual frame deferral would actually take each deferred frame
and queue them onto the channel.
This change in behavior caused the confbridge_recording test to start
failing consistently. Without going too crazily deep into the details,
a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect
was attempting to break it out of the sleep, but because there were more
frames in the channel read queue than expected, the channel ended up
being unable to break from its sleep loop.
By restoring the behavior of individual frame queuing after deferral,
the test starts passing again.
Note, this points to a potential underlying issue pointing to an
"unbalance" that can occur when queuing multiple frames at once,
and so a follow-up issue is being created to investigate that
possibility.
Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d
The sending codec is switched to the receiving codec and then
is switched back to the best native codec on EVERY receiving RTP packets.
This is because after call of ast_channel_set_rawwriteformat there is call
of ast_set_write_format which calls set_format which sets rawwriteformat
to the best native format.
This patch adds a new function ast_set_write_format_path which set
specific write path on channel and uses this function to switch
the sending codec.
ASTERISK-26603 #close
Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d
Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992
that requested ability to add callerid into app_originate.
Comments in that issue suggested that it was better solved by
adding an option to gosub prior to originating the call. The
attached patch implements this much like app_dial with two
options one to gosub on the originating channel and one to gosub
on the newly created channel and behaves just like app_dial.
I have tested this patch by adding callerid info to the new
channel and also SIPAddHeader (to e.g. add header to force auto
answer) and confirmed it works. Have also tested both 'exten'
and 'app' versions of app_originate.
Opened by: dkerr
Patch by: dkerr
Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57
The asterisk.h header file needs to be included first or else
some things go awry, such as:
implicit declaration of function 'vasprintf'
Change-Id: I981dc2a77a1ba791888e4f1726644d4656c0407c
If a TCP/TLS connection was pending (not accepted and not timed out) during
unload of chan_sip, Asterisk would segfault when trying to send a signal to
a thread whose thread ID hadn't been recorded yet. This commit fixes that by
recording the thread ID before calling the blocking connect() syscall.
This was a regression introduced by 776a14386a.
The above wasn't enough to fix the segfault, which was now delayed to the
point where connect() timed out. Therefore, it was necessary to also remove
the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be
used to interruput the connect() syscall.
This was a regression introduced by 5d313f51b9.
ASTERISK-26586 #close
Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b
When retrieving RTCP stats for PJSIP channels, RTT values are unreliable.
RTT calculation is correct, but the data representation isn't. RTT is
represented by a 32-bit fixed-point number with the integer part in the
first 16 bits and the fractional part in the last 16 bits. In order to
get the RTT value, the fractional part is miscalculated, there is an
unnecessary 16 bit shift that causes overflow. Besides this there is
another mistake, when transforming the integer value to the fixed point
fractional part via bitwise operation, that loses precision.
* RTT fractional part is no longer shifted, avoiding overflow.
* RTT fractional part is transformed to its fixed-point value more
precisely.
* Fixed timeval2ntp() and ntp2timeval() second fraction conversions.
* Fixed NTP timestamp report logging. The usec was inexplicably
multiplied by 4096.
ASTERISK-26566 #close
Reported by Hector Royo Concepcion
Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f
Previously, a TLS server socket would only be restarted upon sip reload if the
bind address had changed. This commit adds checking for changes to TLS
parameters like certificate, ciphers, etc. so they get picked up without
requiring a reload of the entire chan_sip module. This does not affect open
connections in any way, but new connections will use the new TLS parameters.
The changes also apply to HTTP and Manager.
ASTERISK-26604 #close
Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6
libasteriskpj was hard coded to use -lrt but librt is linux specific
so we now use the LIB_RT variable which gets set by configure.
Change-Id: I41148884517e3031f7675a413d524c86e8614694
Fix support of OS's like openBSD that use an older nameser.h,
this change reverts the defines to the older style which on other
systems is found in nameser_compat.h
Tested on openBSD 6.0, Debian 8
ASTERISK-26608 #close
Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a
OpenBSD's 'find' doesn't take the -delete argument so you have to pipe
through 'xargs rm -rf'.
'echo -e' doesn't like \t starting a line. It just prints 't' which
causes the libasteriskpj.exports file to be garbage. They were just
cosmetic so they were removed.
librt doesn't exist so the link of libasteriskpj.so fails. It's not
actually needed for linux anyway so -lrt was removed from the link.
res_rtp_asterisk was failing to load because of an undefined
DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if
so DTLSv1_method is used instead.
ASTERISK-26608
Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c
When Opus is negotiated but not loaded, the log is spammed with messages
because the system does not know how to calculate the number of samples in
a frame.
* Suppress the warning by supplying a function that assumes 20ms of
samples in the frame. For pass through support it doesn't really seem to
matter what number of samples is returned anyway.
ASTERISK-26605 #close
Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f
One of the code paths in __ast_file_read_dirs will only get executed if
the OS doesn't support dirent->d_type OR if the filesystem the
particular file is on doesn't support it. So, while standard Linux
systems support the field, some filesystems like XFS do not. In this
case, we need to call stat() to determine whether the directory entry
is a file or directory so we append the filename to the supplied
directory path and call stat. We forgot to truncate path back to just
the directory afterwards though so we were passing a complete file name
to the callback in the dir_name parameter instead of just the directory
name.
The logic has been re-written to only create a full_path if we need to
call stat() or if we need to descend into another directory.
Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba
fopencookie/funclose is a non-standard API and should not be used
in portable software. Additionally, the way FILE's fd is used in
non-blocking mode is undefined behaviour and cannot be relied on.
This introduces internal abstraction for io streams, that allows
implementing the desired virtualization of read/write operations
with necessary timeout handling.
ASTERISK-24515 #close
ASTERISK-24517 #close
Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85
In multi-party bridges, Asterisk currently supports two video modes:
* Follow the talker, in which the speaker with the most energy is shown
to all participants but the speaker, and the speaker sees the
previous video source
* Explicitly set video sources, in which all participants see a locked
video source
Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.
This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
Removes any explicit video source, and sets the video mode to talk
detection
ASTERISK-26595 #close
Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621
