In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
On 2's compliment machines abs(INT_MIN) behavior is undefined and
results in a negative value still being returnd. This results in
negative hash codes that can result in crashes.
ASTERISK-26528 #close
Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b
0035-r5572-svn-backport-dialog-transaction-deadlock.patch
0036-r5573-svn-backport-ua-pjsua-transaction-deadlock.patch
0037-r5576-svn-backport-session-timer-crash.patch
Also removed the progress bar from wget download to stdout.
ASTERISK-26905 #close
Reported-by: Ross Beer
Change-Id: I268fb3cf71a3bb24283ff0d24bd8b03239d81256
* Report failures if configure finds a required header is missing.
* Deduplicate includes between asterisk.h, astmm.h and compat.h.
* Unconditionally include headers in compat.h if required elsewhere.
Change-Id: Ie67d0185ca71fbfb81c9bdfaebe46a49e3c56dc5
The ast_json structure is used in many Asterisk headers and is often the
only part of json.h used. This adds a forward declaration to asterisk.h
and removes the include of json.h from many headers. The declaration
has been left in endpoints.h and stasis.h to avoid problems with source
files that use ast_json functions without directly including json.h.
ari.h continues to include json.h as it uses enum
ast_json_encoding_format.
Change-Id: Id766aabce6bed56626d27e8d29f559b5e687b769
This change cleans up state management for media streams by moving
RTP instances into their own session structure and adding additional
details that are not relevant to the core (such as connection address).
These can live either in the local capabilities or joint capabilities.
The ability to set explicit connection address information for
the purposes of direct media and NAT has also been added at the
global and stream specific level.
ASTERISK-26900
Change-Id: If7e5307239a9534420732de11c451a2705b6b681
The ao2_global_obj_release() function holds an exclusive lock on the
global object while it is being dereferenced. Any destructors that
run during this time that call ao2_global_obj_ref() will deadlock
because a read lock is required.
Instead, we make the global object inaccessible inside of the write
lock and only dereference it once we have released the lock. This
allows the affected destructors to fail gracefully.
While this doesn't completely solve the referenced issue (the error
message about not being able to create an IQ continues to be shown)
it does solve the backtrace spew that accompanied it.
ASTERISK-21009 #close
Reported by: Marcello Ceschia
Change-Id: Idf40ae136b5070dba22cb576ea8414fbc9939385
This change removes the old epoll support which has not been used or
maintained in quite some time.
The fixed number of file descriptors on a channel has also been removed.
File descriptors are now contained in a growable vector. This can be
used like before by specifying a specific position to store a file
descriptor at or using a new API call, ast_channel_fd_add, which adds
a file descriptor to the channel and returns its position.
Tests have been added which cover the growing behavior of the vector
and the new API call.
ASTERISK-26885
Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928
Updated the AMI version for the following reason (see CHANGES for more details):
The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now
contains a new optional parameter, 'MatchHeader'.
Change-Id: Ie206913ef1dcfa6a2ebe3282da2387e52d6f05b9
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
Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.
ASTERISK-26864
Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6
Rather than hard-coding UDP, allow consumers of the HEP API to specify
which protocol is in use. Update the PJSIP provider to pass in the
current protocol type.
ASTERISK-26850 #close
Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978
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
A new transport parameter 'symmetric_transport' has been added.
When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output. On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.
* config_transport was modified to accept and store the new parameter.
* config_transport/transport_apply was updated to store the transport
name in the pjsip_transport->info field using the pjsip_transport->pool
on UDP transports.
* A 'multihomed_on_rx_message' function was added to
pjsip_message_ip_updater that, for incoming requests, retrieves the
transport name from pjsip_transport->info and retrieves the transport.
If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
containing the transport name is added to the incoming Contact header.
* An 'ast_sip_get_transport_name' function was added to res_pjsip.
It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
transport name if endpoint->transport is set or if there's an
'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
ipv6 address. Otherwise it returns NULL.
* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
pjsip_tpselector. It calls ast_sip_get_transport_name() and if
a non-NULL is returned, sets the selector and sets the transport
on the dialog. If a selector was passed in, it's updated.
* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
were modified to call ast_sip_dlg_set_transport() instead of their
original logic.
* res_pjsip/create_out_of_dialog_request was modified to call
ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
instead of its original logic.
* Existing transport logic was removed from endpt_send_request
since that can only be called after a create_out_of_dialog_request.
* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
a new 'ast_sip_create_rdata_with_contact' function which allows
a contact_uri to be specified in addition to the existing
parameters. (See below)
* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
since all it did was transport selection and that is now done in
ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.
* 'contact_uri' was added to subscription_persistence. This was
necessary because although the parsed rdata contact header has the
x-ast-txp parameter added (if appropriate),
subscription_persistence_update stores the raw packet which
doesn't have it. subscription_persistence_recreate was then
updated to call ast_sip_create_rdata_with_contact with the
persisted contact_uri so the recreated subscription has the
correct transport info to send the NOTIFYs.
* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
all it did was transport selection and that is now done in
ast_sip_create_dialog_uac.
* pjsip_message_ip_updater/multihomed_on_tx_message was updated
to remove all traces of the x-ast-txp parameter from the
outgoing headers.
NOTE: This change does NOT modify the behavior of permanent
contacts specified on an aor. To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated. If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.
You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.
Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f
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
This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.
A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.
The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.
ASTERISK-26732 #close
Reported by Dan Jenkins
Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5
* 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 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
* 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
The bridge_native_rtp module did not properly handle the case where
a smart bridge operation occurs while a channel is suspended. In this
scenario the module would incorrectly set up local or remote RTP
bridging despite the media having to flow through Asterisk. The remote
endpoint would see two media streams and experience wonky audio.
The module has been changed so that it ensures both channels are
not suspended when performing the native RTP bridging and this
requirement has been documented in the bridge technology.
ASTERISK-26781
Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c
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
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
All of the realtime backends create artificial ast_categorys to pass
back into the core as query results. These categories have no filename
or line number information associated with them and the backends differ
slightly on how they create them. So create a couple helper macros to
help make things more consistent.
Also updated the call sites to remove redundant error messages about
memory allocation failure.
Note that res_config_ldap sets the category filename to the 'table name'
but that is not read by anything in the core, so I've dropped it.
Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897
