An #include right after a [section](+) would associate any variable
assignments before a new section in the #include with the wrong section.
* Fix section association by setting the current section to the appended
section.
* Fix '+' and '!' section flag interaction corner case depending upon
which flag came first. If the '!' came first then it would be ignored.
If the '!' came after then it would affect the appended section. The '!'
will now no longer be ignored.
ASTERISK-25461 #close
Reported by: Sean Pimental
Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3
When GetConfigJSON was introduced back in 1.6, it returned each
section as an array of strings: ["key=value", "key2=value2"].
Afterwards, it was changed a few times and became
["key": "value", "key2": "value2"], which is not a correct JSON.
This patch fixes that by constructing a JSON object {} instead of
an array [].
Also, the keys "istemplate" and "tempates" that are used to
indicate templates and their inherited categories are now wrapped in
quotes.
ASTERISK-25391 #close
Reported by: Bojan Nemčić
Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8
When Asterisk is part of a larger distributed system, log files are often
gathered using tools (such as logstash) that prefer to consume information
and have it rendered using other tools (such as Kibana) that prefer a
structured format, e.g., JSON. This patch adds support for JSON formatted
logs by adding support for an optional log format specifier in Asterisk's
logging subsystem. By adding a format specifier of '[json]':
full => [json]debug,verbose,notice,warning,error
Log messages will be output to the 'full' channel in the following
format:
{
"hostname": Hostname or name specified in asterisk.conf
"timestamp": Date/Time
"identifiers": {
"lwp": Thread ID,
"callid": Call Identifier
}
"logmsg": {
"location": {
"filename": Name of the file that generated the log statement
"function": Function that generated the log statement
"line": Line number that called the logging function
}
"level": Log level, e.g., DEBUG, VERBOSE, etc.
"message": Actual text of the log message
}
}
ASTERISK-25425 #close
Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238
There was a problem observed where the "logger add channel" CLI command
would allow for a channel with the same name to be added multiple times.
This would result in each message being written out to the same file
multiple times.
The problem was due to the difference in how logger channel filenames
are stored versus the format they are allowed to be presented when they
are added. For instance, if adding the logger channel "foo" through the
CLI, the result would be a logger channel with the file name
/var/log/asterisk/foo being stored. So when trying to add another "foo"
channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily
add the duplicate channel.
The fix presented here is to introduce two new methods in the logger
code:
* make_filename(): given a logger channel name, this creates the
filename for that logger channel.
* find_logchannel(): given a logger channel name, this calls
make_filename() and then traverses the list of logchannels in order
to find a match.
This change has made use of make_filename() and find_logchannel()
throughout to more consistently behave.
ASTERISK-25305 #close
Reported by Mark Michelson
Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36
When leaving a bridge, indications on a channel could be swallowed by
the internal indication logic because it appears that the channel is on
its way to be hung up anyway. One such situation where this is
detrimental is when channels on hold are redirected out of a bridge. The
AST_CONTROL_UNHOLD indication from the bridging code is swallowed,
leaving the channel in question to still appear to be on hold.
The fix here is to modify the logic inside ast_indicate_data() to not
drop the indication if the channel is simply leaving a bridge. This way,
channels on hold redirected out of a bridge revert to their expected "in
use" state after the redirection.
ASTERISK-25418 #close
Reported by Mark Michelson
Change-Id: If6115204dfa0551c050974ee138fabd15f978949
Page uses the async method of dialing with the dial API. When a call gets
forwarded there is no calling channel available. If the predial handler
was set then the calling channel could not be put into auto-service
for the forwarded call because it doesn't exist. A crash is the result.
* Moved the callee predial parameter string processing to before the
string is passed to the dial API rather than having the dial API do it.
There are a few benefits do doing this. The first is the predial
parameter string processing doesn't need to be done for each channel
called by the dial API. The second is in async mode and the forwarded
channel is to have the predial handler executed on it then the
non-existent calling channel does not need to be present to process the
predial parameter string.
* Don't start auto-service on a non-existent calling channel to execute
the predial handler when the dial API is in async mode and forwarding a
call.
ASTERISK-25384 #close
Reported by: Chet Stevens
Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981
Currently, Asterisk will log to the last configured syslog
channel in logger.conf. This is due to the fact that the
final call to openlog() supersedes all of the previous calls.
This commit removes the call to openlog() and passes the
facility to ast_log_vsyslog(), along with utilizing the
LOG_MAKEPRI macro to ensure that the message is routed to
the correct facility and with the correct priority.
ASTERISK-25407 #close
Reported by: Elazar Broad
Tested by: Elazar Broad
Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2
This patch adds support for receiving events regarding Peer status changes
and Contact status changes. This is particularly useful in scenarios where
we are subscribed to all endpoints and channels, where we often want to know
more about the state of channel technology specific items than a single
endpoint's state.
ASTERISK-24870
Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9
Now with menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", a timestamp is
shown with each file descriptor. This helps to debug leaked UDP/TCP ports on
long-lived servers, for example.
ASTERISK-25405 #close
Change-Id: I968339e5155a512eba1032a5263f1ec8b5e1f80b
When changing a hint extension without removing the hint first the
device state and presence state is not updated. This causes the state
of the hint to be that of the previous extension and not the current
one. This state is kept until a state change occurs as a result of
something (presence state change, device state change).
This change updates the hint with the current device and presence
state of the new extension when it is changed. Any state callbacks
which may have been added before the hint extension is changed are
also informed of the new device and presence state if either have
changed.
ASTERISK-25394 #close
Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f
When Asterisk translates between codecs, each with a different frame size (for
example between iLBC 30 and Speex-WB), too large frames were created by
ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame
length, creating several frames when necessary. Affects all transcoding modules
which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex.
ASTERISK-25353 #close
Change-Id: I2e229569d73191d66a4e43fef35432db24000212
It has been observed that on long-running busy systems, a scheduler
context can eventually hit INT_MAX for its assigned IDs and end up
overflowing into a very low negative number. When this occurs, this can
result in odd behaviors, because a negative return is interpreted by
callers as being a failure. However, the item actually was successfully
scheduled. The result may be that a freed item remains in the scheduler,
resulting in a crash at some point in the future.
The scheduler can overflow because every time that an item is added to
the scheduler, a counter is bumped and that counter's current value is
assigned as the new item's ID.
This patch introduces a new method for assigning scheduler IDs. Instead
of assigning from a counter, a queue of available IDs is maintained.
When assigning a new ID, an ID is pulled from the queue. When a
scheduler item is released, its ID is pushed back onto the queue. This
way, IDs may be reused when they become available, and the growth of ID
numbers is directly related to concurrent activity within a scheduler
context rather than the uptime of the system.
Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2
Asterisk can load and register an object type while still having an invalid
sorcery mapping. This can cause an issue when a creation call is invoked.
For example, mis-configuring PJSIP's endpoint identifier by IP address mapping
in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a
subsequent ARI invocation to create the object will cause a crash, as the
internal type may not be registered as sorcery expects.
Merely checking for a NULL pointer here solves the issue.
Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac
In ast_endpoint_snapshot_create(), a failure to init the
string fields results in two attempts to ao2_cleanup the
same pointer. Removed RAII_VAR to eliminate problem.
ASTERISK-25375 #close
Reported by: Scott Griepentrog
Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979
When issuing the "core show hints" CLI command a combination of both
the hint extension and context is created. This uses a fixed size
buffer expecting that the extension will not exceed maximum extension
length. When the extension is actually a pattern match this constraint
does not hold true, and the extension may exceed the maximum extension
length. In this case extra characters are written past the end of the
fixed size buffer.
This change makes it so the construction of the combined hint extension
and context can not exceed the size of the buffer.
ASTERISK-25367 #close
Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499
When unreferencing a taskprocessor its reference count is checked
to determine if it should be unlinked from the taskprocessors
container and its listener shut down. In between the time when the
reference count is checked and unlinking it is possible for
another thread to jump in, find it, and get a reference to it. If
the thread then uses the taskprocessor it may find that it is not
in the state it expects.
This change locks the taskprocessors container during almost the
entire unreference operation to ensure that any other thread which
may attempt to find the taskprocessor has to wait.
ASTERISK-25295
Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c
When deleting a scheduled item if the item in question is currently
executing the ast_sched_del function waits until it has completed.
This is accomplished using ast_cond_wait. Unfortunately the
ast_cond_wait function can suffer from spurious wakeups so the
predicate needs to be checked after it returns to make sure it has
really woken up as a result of being signaled.
This change adds a loop around the ast_cond_wait to make sure that
it only exits when the executing task has really completed.
ASTERISK-25355 #close
Change-Id: I51198270eb0b637c956c61aa409f46283432be61
When executing an action in a bridge it is possible for the
channel to be hung up without the bridge becoming aware of it.
This is most easily reproducible by hanging up when the bridge
is streaming DTMF due to a feature timeout. This change makes
it so after action execution the channel is checked to determine
if it has been hung up and if it has it is kicked from the bridge.
ASTERISK-25341 #close
Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062
* Make ast_rtp_codecs_payload_code() get the current mapping or create a
rx payload type mapping.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: Ia4b2d45877a8f004f6ce3840e3d8afe533384e56
There are numerous problems with the current implementation of the RTP
payload type mapping in Asterisk. It uses only one mapping structure to
associate payload types to codecs. The single mapping is overkill if all
of the payload type values are well known values. Dynamic payload type
mappings do not work as well with the single mapping because RFC3264
allows each side of the link to negotiate different dynamic mappings for
what they want to receive. Not only could you have the same codec mapped
for sending and receiving on different payload types you could wind up
with the same payload type mapped to different codecs for each direction.
1) An independent payload type mapping is needed for sending and
receiving.
2) The receive mapping needs to keep track of previous mappings because of
the slack to when negotiation happens and current packets in flight using
the old mapping arrive.
3) The transmit mapping only needs to keep track of the current negotiated
values since we are sending the packets and know when the switchover takes
place.
* Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
use the new function because ast_rtp_codecs_payload_code() was used for
mappings in both directions.
* Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
to pass preferred codec mappings to the peer channel for early media
bridging or when we need to prefer the offered mapping that RFC3264 says
we SHOULD use.
* ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
the only new public functions created. All the others were only used for
the tx or rx mapping direction so the function doxygen now reflects which
direction the function operates.
* chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
that makes no sense when processing an incoming SDP. We would be wiping
out any mappings that we set for the possible outgoing SDP we sent
earlier.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac
When allocating a sorcery object, fail if the
id value was not allocated.
ASTERISK-25323
Reported by: Scott Griepentrog
Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e
The built frame format in audiohook_read_frame_both() is now set to a
signed linear format before the rx and tx frames are duplicated instead of
only for the mixed audio frame duplication.
ASTERISK-25322 #close
Reported by Sean Pimental
Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538
Some codecs that may be a third party library to Asterisk need to have
knowledge of the format attributes that were negotiated. Unfortunately,
when the great format migration of Asterisk 13 occurred, that ability
was lost.
This patch adds an API call, ast_format_attribute_get, to the core
format API, along with updates to the unit test to check the new API
call. A new callback is also now available for format attribute modules,
such that they can provide the format attribute values they manage.
Note that the API returns a void *. This is done as the format attribute
modules themselves may store format attributes in any particular manner
they like. Care should be taken by consumers of the API to check the
return value before casting and dereferencing. Consumers will obviously
need to have a priori knowledge of the type of the format attribute as
well.
Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3
clock_gettime() is, unfortunately, not portable. But I did like that
over our usual `ts.tv_nsec = tv.tv_usec * 1000` copy/paste code we
usually do when we want a timespec and all we have is ast_tvnow().
This patch adds ast_tsnow(), which mimics ast_tvnow(), but returns a
timespec. If clock_gettime() is available, it will use that. Otherwise
ast_tsnow() falls back to using ast_tvnow().
Change-Id: Ibb1ee67ccf4826b9b76d5a5eb62e90b29b6c456e
An http request can be sent to get the existing Asterisk logs.
The command "curl -v -u user:pass -X GET 'http://localhost:8088
/ari/asterisk/logging'" can be run in the terminal to access the
newly implemented functionality.
* Retrieve all existing log channels
ASTERISK-25252
Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808
An http request can be sent to create a log channel
in Asterisk.
The command "curl -v -u user:pass -X POST
'http://localhost:088/ari/asterisk/logging/mylog?
configuration=notice,warning'" can be run in the terminal
to access the newly implemented functionality for ARI.
* Ability to create log channels using ARI
ASTERISK-25252
Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782
An http request can be sent to delete a log channel
in Asterisk.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/logging/mylog'" can be run in the terminal
to access the newly implemented functionally for ARI.
* Able to delete log channels using ARI
ASTERISK-25252
Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6
An http request can be sent to rotate a specified log channel.
If the channel does not exist, an error response will be
returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/logging/logChannelName/rotate'" can be run in the
terminal to access this new functionality.
* Added the ability to rotate log files through ARI
ASTERISK-25252
Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01
A testsuite test recently failed due to a crash that occurred in the DNS
core. The problem was that the test could not resolve an address, did
not set a result on the DNS query, and then indicated the query was
completed. The DNS core does not handle the case of a query with no
result gracefully, and so there is a crash.
This changeset makes the DNS system resolver set a result with a
zero-length answer in the case that a DNS resolution failure occurs
early. The DNS core now also will accept such a response without
treating it as invalid input. A unit test was updated to no longer treat
setting a zero-length response as off-nominal.
Change-Id: Ie56641e22debdaa61459e1c9a042e23b78affbf6
ast_rtp_codecs_get_payload() gets called once or twice for every received
RTP frame so it would be nice to not allocate an ao2 object to then have
it destroyed shortly thereafter. The ao2 object gets allocated only if
the payload type is not set by the channel driver as a negotiated value.
The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323.
* Made static_RTP_PT[] an array of ao2 objects that
ast_rtp_codecs_get_payload() can return instead of an array of structs
that must be copied into a created ao2 object.
ASTERISK-25296 #close
Reported by: Richard Mudgett
Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0
* Fix off nominial ref leak of new_type in
ast_rtp_codecs_payloads_set_m_type().
* No need to lock static_RTP_PT_lock in
ast_rtp_codecs_payloads_set_m_type() and
ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type
parameter sanity check.
* No need to create ast_rtp_payload_type ao2 objects with a lock since the
lock is not used.
Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4
This event is necessary for the bridge_wait_e_options test to be able to
confirm that ringing is being played on the local channel that runs the
BridgeWait() application with the e(r) option.
ASTERISK-25292 #close
Reported by Kevin Harwell
Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e
Currently, if a blank musiconhold.conf is used, musiconhold will fail
to start for a channel going into a holding bridge with an anticipation
of getting music on hold. That being the case, no frames will be written
to the channel and that can pose a problem for blind transfers in PJSIP
which may rely on frames being written to get past the REFER framehook.
This patch makes holding bridges start a silence generator if starting
music on hold fails and makes it so that if no music on hold functions
are installed that the ast_moh_start function will report a failure so
that consumers of that function will be able to respond appropriately.
ASTERISK-25271 #close
Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99
(cherry picked from commit 8458b8d441)
This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
Commit 54b25c80c8 solved an issue where a
specific scenario involving local channels and a native local RTP bridge
could result in ringback still being heard on a calling channel even
after the call is bridged.
That commit caused many tests in the testsuite to fail with alarming
consequences, such as not sending DialBegin and DialEnd events, and
giving incorrect hangup causes during calls.
This commit reverts the previous commit and implements and alternate
solution. This new solution involves only passing AST_CONTROL_RINGING
frames across local channels if the local channel is in AST_STATE_RING.
Otherwise, the frame does not traverse the local channels. By doing
this, we can ensure that a playtones generator does not get started on
the calling channel but rather is started on the local channel on which
the ringing frame was initially indicated.
ASTERISK-25250 #close
Reported by Etienne Lessard
Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39
Previous changes to sample rate support in audiohooks accidentally
removed code responsible for allowing the manipulate audiohooks
to work. Without this code the manipulated frame would be dropped
and not used. This change restores it.
ASTERISK-25253 #close
Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13
Control frames with a -1 payload are used as a special signal to stop
playtones generators on channels. This indication is sent both by
app_dial as well as by ast_answer() when a call is answered in case any
tones were being generated on a calling channel.
This control frame type was made to stop traversing local channel pairs
as an optimization, because it was thought that it was unnecessary to
send these indications, and allowing such unnecessary control frames to
traverse the local channels would cause the local channels to optimize
away less quickly.
As it turns out, through some special magic dialplan code, it is
possible to have a tones being played on a non-local channel, and it is
important for the local channel to convey that the tones should be
stopped. The result of having tones continue to be played on the
non-local channel is that the tones play even once the channel has been
bridged. By not blocking the -1 control frame type, we can ensure that
this situation does not happen.
ASTERISK-25250 #close
Reported by Etienne Lessard
Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815
Due to changes in audiohooks to support different sample rates the
underlying storage of samples is in the format of the audiohook
itself and not of the format being requested. This means that if a
channel is using G722 the samples stored will be at 16kHz. If
something subsequently reads from the audiohook at a format which
is not the same sample rate as the audiohook the number of samples
needs to be adjusted.
Given the following example:
1. Channel writing into audiohook at 16kHz (as it is using G722).
2. Chanspy reading from audiohook at 8kHz.
The original code would read 160 samples from the audiohook for
each 20ms of audio. This is incorrect. Since the audio in the
audiohook is at 16kHz the actual number needing to be read is 320.
Failure to read this much would cause the audiohook to reset
itself constantly as the buffer became full.
This change adjusts the requested number of samples by determining
the duration of audio requested and then calculating how many
samples that would be in the audiohook format.
ASTERISK-25247 #close
Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d
This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
chan_sip option, this specifies an interval, in seconds, at which we
will send RTP comfort noise frames. This can be useful for keeping RTP
sessions alive as well as keeping NAT associations alive during lulls.
ASTERISK-25242 #close
Reported by Mark Michelson
Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d
Fixes for issues with the ASTERISK-24934 patch.
* Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is
an empty string. If it were an empty string the functions returned NULL
as if there were a memory allocation failure. This failure caused the AMI
VarSet event to not get posted if the new value was an empty string.
* Fixed dest buffer overwrite potential in ast_escape() and
ast_escape_c(). If the dest buffer size is smaller than the space needed
by the escaped s parameter string then the dest buffer would be written
beyond the end by the nul string terminator. The num parameter was really
the dest buffer size parameter so I renamed it to size.
* Made nul terminate the dest buffer if the source string parameter s was
an empty string in ast_escape() and ast_escape_c().
* Updated ast_escape() and ast_escape_c() doxygen function description
comments to reflect reality.
* Added some more unit test cases to /main/strings/escape to cover the
empty source string issues.
ASTERISK-25255 #close
Reported by: Richard Mudgett
Change-Id: Id77fc704600ebcce81615c1200296f74de254104
An http request can be sent to retrieve information on a single
module, including the resource name, description, use count, status,
and support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on a single module can now be retrieved
ASTERISK-25173
Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463
During an attended transfer a thread is started that handles imparting the
bridge channel. From the start of the thread to when the bridge channel is
ready exists a gap that can potentially cause problems (for instance, the
channel being swapped is hung up before the replacement channel enters the
bridge thus stopping the transfer). This patch adds a condition that waits
for the impart thread to get to a point of acceptable readiness before
allowing the initiating thread to continue.
ASTERISK-24782
Reported by: John Bigelow
Change-Id: I08fe33a2560da924e676df55b181e46fca604577
This patch adds five CLI commands for the media cache:
* 'media cache show all' - display a summary of all items in the media
cache.
* 'media cache show <uri>' - display detailed information about a
single item in the media cache.
* 'media cache delete <uri>' - remove an item from the media cache, and
inform the bucket backend for the URI scheme to remove the item as
well.
* 'media cache refresh <uri>' - refresh a URI. If the item does not
exist in the media cache, the bucket backend will pull down the media
associated with the URI and create the item in the cache.
* 'media cache create <uri>' - create an item in the media cache from
some local media storage. Note that the bucket backend for the URI
scheme must still permit the item creation.
Change-Id: Id1c5707a3b8e2d96b56e4691a46a936cd171f4ae
This patch adds a new API to the Asterisk core that acts as a media
cache. The core API itself is mostly a thin wrapper around some bucket
API provided implementation that itself acts as the mechanism of
retrieval for media. The media cache API in the core provides the
following:
* A very thin in-memory cache of the active bucket_file items. Unlike a
more traditional cache, it provides no expiration mechanisms. Most
queries that hit the in-memory cache will also call into the bucket
implementations as well. The bucket implementations are responsible
for determining whether or not the active record is active and valid.
This makes sense for the most likely implementation of a media cache
backend, i.e., HTTP. The HTTP layer itself is the actual arbiter of
whether or not a record is truly active; as such, the in-memory cache
in the core has to defer to it.
* The ability to create new items in the media cache from local
resources. This allows for re-creation of items in the cache on
restart.
* Synchronization of items in the media cache to the AstDB. This
also includes various pieces of important metadata.
The API provides sufficient access that higher level APIs, such as the
file or app APIs, do not have to worry about the semantics of the bucket
APIs when needing to playback a resource.
In addition, this patch provides unit tests for the media cache API. The
unit tests use a fake bucket backend to verify correctness.
Change-Id: I11227abbf14d8929eeb140ddd101dd5c3820391e
This patch adds a new function to the bucket API for ast_bucket_file
objects, ast_bucket_file_metadata_callback. It will call ao2_callback on
the ast_bucket_file's ao2_container of metadata, calling the provided
ao2_callback_fn callback on each piece of metadata associated with the
file.
This is particularly useful when a bucket backend has added metadata,
and a higher level API wants to be aware of/access said metadata,
without knowing for sure what the key is.
Change-Id: I96f6757717f47b650df91a437f7df16406227466
Some individual fields may fail their conversion due to their default
values being invalid for their custom handlers. In particular,
configuration values that depend on others being enabled (and thus have
an empty default value) are notorious for tripping this routine up. An
example of this are any of the DTLS options for endpoints. Any of the
DTLS options will fail to be applied (as DTLS is not enabled), causing
the entire object set to be aborted.
This patch makes it so that we log a debug message when skipping a
field, and rumble on anyway.
ASTERISK-25238
Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76
We have a strange relationship between the parsing of format
capabilities from a string and their representation as a string. We
expect the format capabilities to be expressed as a string in the
following format:
allow = !all,ulaw,alaw
disallow = g722
While we would generate the string representation of those formats as:
allow = (ulaw|alaw)
disallow = (ulaw|alaw|g729...)
When the configuration framework needs to store values as a string, it
generates the format capabilities using the second representation; this
representation however cannot be parsed when the entry is rehydrated.
This patch fixes that by updating
ast_format_cap_update_by_allow_disallow to parse an entry as if it were
in the generated format if it has a leading '(' and a trailing ')'.
ASTERISK-25238
Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca
If a sorcery wizard does not support one of the 'optional' CRUD
operations (namely the CUD), log a WARNING message so we are aware of
why the operation failed. This also removes an assert in this case, as
the CUD operation may have been triggered by an external system, in
which case it is not a programming error but a configuration error.
Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53
Currently, the device state provider API will allow you to register a
device state provider with the same case insensitive name more than
once. This could cause strange issues, as the duplicate device state
providers will not be queried when a device's state has to be polled.
This patch updates the API such that a device state provider with the
same name as one that has already registered will be rejected.
Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2
An http request can be sent to retrieve a list of all existing modules,
including the resource name, description, use count, status, and
support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
asterisk/modules" (or something similar, depending on configuration)
can be run in the terminal to access this new functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on modules can now be retrieved
Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0
Prior to this patch, the DNS core present in master had no default system-level
resolver implementation. Therefore, it was not possible for the DNS core to
perform resolutions unless the libunbound library was installed and the
res_resolver_unbound module was loaded.
This patch introduces a system-level DNS resolver implementation that will
register itself with the lowest consideration priority available (to ensure
that it is to be used only as a last resort). The resolver relies on low-level
DNS search functions to perform a rudimentary DNS search based on a provided
query and then supplies the search results to the DNS core.
ASTERISK-25146 #close
Reported By: Joshua Colp
Change-Id: I3b36ea17b889a98df4f8d80d50bb7ee175afa077
This patch enhances the bucket API in two ways.
First, since ast_bucket and ast_bucket_file instances are immutable, a 'clone'
operation has been added that provides a 'clone' of an existing
ast_bucket/ast_bucket_file object. Note that this makes use of the
ast_sorcery_copy operation, along with the copy callback handler on the
"bucket" and "file" object types for the bucket sorcery instance.
Second, there is a need for the bucket API to ask a wizard if an object
is stale. This is particularly useful with the upcoming media cache
enhancements, where we want to ask the backing data storage if the
object we are currently operating on has known updates. This patch adds
API calls for ast_bucket and ast_bucket_file objects, which callback
into their respective sorcery wizards via the sorcery API.
Unit tests have also been added to cover the respective
ast_bucket/ast_bucket_file clone and staleness operations.
Change-Id: Ib0240ba915ece313f1678a085a716021d75d6b4a
This patch enhances the sorcery API to allow for sorcery wizards to
determine if an object is stale. This includes the following:
* Sorcery objects now have a timestamp that is set on creation. Since
sorcery objects are immutable, this can be used by sorcery wizards to
determine if an object is stale.
* A new API call has been added, ast_sorcery_is_stale. This API call
queries the wizards associated with the object, calling a new callback
function 'is_stale'. Note that if a wizard does not support the new
callback, objects are always assumed to not be stale.
* Unit tests have been added that cover the new API call.
Change-Id: Ica93c6a4e8a06c0376ea43e00cf702920b806064
When a mapping does not exist between a sorcery.conf defined object and
a realtime mapping in extconf, currently, the user will receive a slew
of ERROR messages that don't really tell what is happening. Some ERROR
messages may even be misleading, as they occur after the sorcery API has
already given up on the attempt to load and create the sorcery object.
This patch adds a bit of debug and a useful WARNING message for when a
wizard's open callback fails for a particular object type. In the bad
configurations that resulted in this patch, this provided a 'root cause'
WARNING message that pointed in the right direction of the configuration
problem.
Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b
The recurring unit test expects the user data on a DNS query
created as a result of a recurring DNS query to be the recurring
structure itself. This is true, mostly. When invoking the user
provided callback this user data is changed to the user provided
data. This presents a race condition where the data may or may
not point to the recurring data.
This change simplifies the callback of the user provided callback
by creating a new query and populating it with the expected values.
This leaves the recurring DNS query alone and fixes the race
condition. This is more in line with how the API should be used
overall.
ASTERISK-25222 #close
Change-Id: I10fb6deec025dff097157e7ec17e6e4921778478
When running valgrind on Asterisk, it complained about:
==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304)
==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...)
==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292)
==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437)
The code in question is a struct assignment, which may be performed by
memcpy as a compiler optimization. It is changed to only copy the struct
contents if source and destination are different.
ASTERISK-25219 #close
Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a
If DEBUG_FD_LEAKS was used and more file descriptors than the default of
1024 were available, some DEBUG_FD_LEAKS-patched functions would
overwrite memory past the fixed-size (1024) fdleaks buffer.
This change:
- adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe
- consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024
- stores pointers to constants instead of copying the contents
- reorders the fdleaks struct for possibly tighter packing
- adds a tiny bit of documentation
ASTERISK-25212 #close
Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5
When a frame is queued on a channel, any failure in
ast_channel_alert_write is logged along with errno.
This change improves the diagnostic message through
aligning the errno value with actual failure cases.
ASTERISK-25224
Reported by: Andrey Biglari
Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b
When 8297136f was merged for ASTERISK-25040, a regression was introduced
surrounding the case sensitivity of device names within hints.
Previously, device names - such as 'sip/foo' - were compared in a case
insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After
that patch, only the case sensitive name would match, i.e., 'SIP/foo'.
As a result, some dialplan hints stopped working.
This patch re-introduces case insensitive matching for device names in
hints.
ASTERISK-25040
ASTERISK-25202 #close
Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c
(cherry picked from commit 96bbcf495a)
Richard Mudgett
Ivan Poddubny
Matt Jordan
Joshua Colp
Mark Michelson
Elazar Broad
Alexander Traud
Scott Griepentrog
Guido Falsi
David M. Lee
Scott Emidy
Benjamin Ford
Jonathan Rose
Kevin Harwell
Ashley Sanders
Walter Doekes