Mark Michelson [Wed, 29 Jul 2015 17:58:23 +0000 (12:58 -0500)]
dns_core: Allow zero-length DNS responses.
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.
Jonathan Rose [Thu, 16 Jul 2015 17:16:25 +0000 (12:16 -0500)]
holding_bridge: ensure moh participants get frames
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.
pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
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.
Mark Michelson [Thu, 23 Jul 2015 18:11:42 +0000 (13:11 -0500)]
Local channels: Alternate solution to ringback problem.
Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d 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.
audiohook: Use manipulated frame instead of dropping it.
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.
Mark Michelson [Wed, 22 Jul 2015 14:46:29 +0000 (09:46 -0500)]
Local channels: Do not block control -1 payloads.
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.
audiohook: Read the correct number of samples based on audiohook format.
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.
cdr/cdr_adaptive_odbc.c: Fix quoted identifier usage when inserting CDR records
Commit a24ce38 added support for the use of quoted indentifiers when inserting
CDR records into the database. However, the if statement logic responsible for
determining whether to use those identifiers is reversed, resulting in a
reference to the quoted identifier character buffer which will be null, hence
null terminating the SQL query, resulting in a truncated statement which
fails to execute.
Rusty Newton [Mon, 20 Jul 2015 17:39:48 +0000 (12:39 -0500)]
Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c
* In sip.conf.sample fix sentence where we said that WS or WSS are supported
transports for use in an outbound register definition. They are not
supported in that case.
* In func_cdr.c made it clear that the Disable option for CDR_PROP can be used
to enable CDR on a channel.
Mark Michelson [Thu, 9 Jul 2015 19:17:53 +0000 (14:17 -0500)]
res_pjsip: Add rtp_keepalive endpoint option.
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.
Matt Jordan [Sun, 19 Jul 2015 14:11:18 +0000 (09:11 -0500)]
res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf
Misconfiguring sorcery.conf with a 'config' wizard with no extra data
will currently crash Asterisk on startup, as the wizard requires a comma
delineated list to parse. This patch updates res_sorcery_config to check
for the presence of the data before it starts manipulating it.
chan_pjsip: Don't change formats when frame of unsupported format is received.
Receipt of an RTP packet currently causes the formats on an PJSIP channel to
change to the format of the RTP packet. In some off-nominal cases it's possible
for this to be a format that has not been configured or negotiated. This change
makes it so only formats explicitly configured on the endpoint are allowed.
Matt Jordan [Wed, 8 Jul 2015 21:39:35 +0000 (16:39 -0500)]
ARI: Add support for push configuration of dynamic object
This patch adds support for push configuration of dynamic, i.e.,
sorcery, objects in Asterisk. It adds three new REST API calls to the
'asterisk' resource:
* GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current
object given its ID. This returns back a list of ConfigTuples, which
define the fields and their present values that make up the object.
* PUT /asterisk/{configClass}/{objectType}/{id}: create or update an
object. A body may be passed with the request that contains fields to
populate in the object. The same format as what is retrieved using
the GET operation is used for the body, save that we specify that the
list of fields to update are contained in the "fields" attribute.
* DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic
object from its backing storage.
Note that the success/failure of these operations is somewhat
configuration dependent, i.e., you must be using a sorcery wizard that
supports the operation in question. If a sorcery wizard does not support
the create or delete mechanisms, then the REST API call will fail with a
403 forbidden.
Richard Mudgett [Wed, 15 Jul 2015 20:40:32 +0000 (15:40 -0500)]
strings.h: Fix issues with escape string functions.
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
Richard Mudgett [Tue, 14 Jul 2015 19:36:42 +0000 (14:36 -0500)]
res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park.
setup_park_common_datastore() was assuming that a non-NULL string returned
for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty
strings. Things got crashy as a result.
* Made setup_park_common_datastore() treat the channel variable values the
same whether they are NULL or empty for ATTENDEDTRANSFER and
BLINDTRANSFER.
ASTERISK-25254 #close
Reported by: Richard Mudgett
Richard Mudgett [Fri, 10 Jul 2015 23:17:52 +0000 (18:17 -0500)]
res_pjsip_session.c: Fix crash on call disconnect.
The crash fix for ASTERISK-25183 backported some code from master to try
to make sure that a BYE response is processed by the same serializer used
by the BYE request. The identified race condition causing that backport
was the BYE request code had not finished processing after sending the BYE
before the BYE response came in for processing under a different thread.
Unfortunately, there is still a race condition. Now the race condition is
between destroying the call session's serializer in
ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a
reference to the serializer for a BYE response. Even worse, the new race
condition is a design limitation of the taskprocessor implementation that
didn't matter in versions before v12. Back then, taskprocessors were only
destroyed when a module unloaded. Now res_pjsip can destroy them when a
call ends.
However, as noted on the ASTERISK-25183 commit,
session_inv_on_state_changed() is disassociating the dialog from the
session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED.
This is a tad too soon because our BYE request transaction has not
completed yet.
* Split session_end() that is called by session_inv_on_state_changed() to
hold off session destruction until the BYE transaction timeout occurs or a
failed initial INVITE transaction timeout occurs in
session_inv_on_tsx_state_changed().
Benjamin Ford [Tue, 14 Jul 2015 18:12:32 +0000 (13:12 -0500)]
ARI: Added new functionality to reload a single module.
An http request can be sent to reload an Asterisk module. If the
module can not be reloaded or is not already loaded, an error
response will be returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, based
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
* Asterisk modules can be reloaded through http requests
Benjamin Ford [Tue, 14 Jul 2015 13:55:14 +0000 (08:55 -0500)]
ARI: Added new functionality to unload a single module.
An http request can be sent to unload an Asterisk module. If the
module can not be unloaded or is already unloaded, an error response
will be returned.
The command "curl -v -u user:pass -X DELETE '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
* Asterisk modules can be unloaded through http requests
Benjamin Ford [Mon, 13 Jul 2015 21:00:19 +0000 (16:00 -0500)]
ARI: Added new functionality to load a single module.
An http request can be sent to load an Asterisk module. If the
module can not be loaded or is loaded already, an error response
will be returned.
The command curl -v -u user:pass -X POST '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
* Asterisk modules can be loaded through http requests
Benjamin Ford [Mon, 13 Jul 2015 15:54:51 +0000 (10:54 -0500)]
ARI: Added new functionality to get information on a single module.
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
Kevin Harwell [Wed, 8 Jul 2015 19:56:10 +0000 (14:56 -0500)]
bridge.c: Fixed race condition during attended transfer
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.
Matt Jordan [Wed, 13 May 2015 21:22:30 +0000 (16:22 -0500)]
media cache: Add CLI commands
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.
Matthew Jordan [Thu, 29 Jan 2015 14:38:23 +0000 (14:38 +0000)]
media cache: Add a core API and facade for a backend agnostic media cache
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.
Matt Jordan [Sun, 12 Jul 2015 01:25:19 +0000 (20:25 -0500)]
main/bucket: Add a callback function for ast_bucket_file objects
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.
Matt Jordan [Wed, 8 Jul 2015 21:28:13 +0000 (16:28 -0500)]
main/sorcery: Don't fail object set creation from JSON if field fails
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.
Matt Jordan [Wed, 8 Jul 2015 21:21:09 +0000 (16:21 -0500)]
main/format_cap: Parse capabilities generated by ast_format_cap_get_names
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:
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 ')'.
Matt Jordan [Wed, 8 Jul 2015 21:38:24 +0000 (16:38 -0500)]
res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails
Having a debug message tell us that we attempted to look up an item but
failed is nice in circumstances when it isn't clear if the wizard was
queried correctly or not.
Matt Jordan [Wed, 8 Jul 2015 21:35:36 +0000 (16:35 -0500)]
res_pjsip/configuration: Fix a variety of default value problems
This patch fixes some bad default value handling in the following
settings:
* The 'message_context' and 'accountcode' settings are not mandatory. As
such, we can allow their stringfield values to be empty.
* The 'media_encryption' setting applies a default value of 'none' to
the setting, which it then can't parse or understand. Since the value
is documented to be 'no', this will now apply that as the default
value.
Matt Jordan [Wed, 8 Jul 2015 21:32:25 +0000 (16:32 -0500)]
main/sorcery: Provide log messages when a wizard does not support an operation
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.
Matt Jordan [Sat, 27 Jun 2015 22:53:37 +0000 (17:53 -0500)]
tests/test_devicestate: Add additional tests for the device state API
This patch adds more tests that exercise the device state API. This includes:
* Tests that cover adding a device state provider, as well as deleting a
device state provider. This also verifies that you cannot add an
already added device state provider, and cannot delete an already
deleted device state provider.
* A test that covers changing device state and receiving said updates
from a device state subscriber. This also covers hitting both the
device state cache as well as a custom device state provider.
* A test that covers converting device state to channel state and device
state values to a string representation and back.
* A test that covers obtaining device state from an active channel and a
channel driver that provides its own device state.
Matt Jordan [Sat, 27 Jun 2015 22:51:43 +0000 (17:51 -0500)]
main/devicestate: Prevent duplicate registration of device state providers
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.
Benjamin Ford [Fri, 26 Jun 2015 15:57:15 +0000 (10:57 -0500)]
ARI: Added new functionality to get all module information.
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
bridge_native_rtp.c: Don't start native RTP bridging after attended transfer.
The bridge_native_rtp module adds a frame hook to channels which are in
a native RTP bridge. This frame hook is used to intercept when a hold
or unhold frame traverses the bridge so native RTP can be stopped or
started as appropriate. This is expected but exposes a specific bug
when attended transfers are involved.
Upon completion of an attended transfer an unhold frame is queued up
to take one of the channels involved off hold. After this is done
the channel is moved between bridges.
When the frame hook is involved in this case for the unhold it
releases the channel lock and acquires the bridge lock. This
allows the bridge core to step in and move the channel
(potentially changing the bridging techology) from another thread.
Once completed the bridge lock is released by the bridge core.
The frame hook is then able to acquire the bridge lock and
wrongfully starts native RTP again, despite the channel no longer
being in the bridge or needing to start native RTP. In fact at
this point the frame hook is no longer attached to the channel.
This change makes it so the native RTP bridge data is available to
the frame hook when it is invoked. Whether the frame hook has
been detached or not is stored on the native RTP bridge data and
is checked by the frame hook before starting or stopping native
RTP bridging. If the frame hook has been detached it does nothing.
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.
res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.
This change moves logic for setting up the DTLS SSL contexts to
when the SDP is done being processed instead of when ICE negotiation
completes. It also stops handshakes from being initiated when we
are acting as a server.
Manipulating the SSL context when ICE negotiation has completed
is problematic as the SSL context is not protected and if acting
as a client the remote side may have started DTLS negotiation
already.
The retransmission timeout timer code has also been split up
and simplified some. Both RTP and RTCP now have their own timers
and the points at which the timer is stopped and started is now
more specific. When a packet is sent the timer is started. When
a response is received but before it is processed the timer is
stopped. This provides a guarantee that the timeout is not
occurring while the response is processed.
MWI subscriptions can crash or corrupt memory when using the subscription
datastore to access the MWI subscription object because the datastore is
not holding a reference to the object.
* Give the subscription datastore a ref to the MWI subscription object.
It is unfortunate that the ref causes a circular ref chain that must be
explicitly broken to allow the memory to get released. The loop is broken
when the subscription is shutdown and if the subscription setup fails.
When res_pjsip body generator modules were generating XML or XPIDF
response bodies, there was a chance that the generated body would be the
exact size of the supplied buffer. Adding the nul string terminator would
then write beyond the end of the buffer and potentially corrupt memory.
* Fix MALLOC_DEBUG high fence violations caused by adding a nul string
terminator on the end of a buffer for XML or XPIDF response bodies.
* Made calls to pj_xml_print() safer if the XML prolog is requested. Due
to a bug in pjproject, the return value could be -1 _or_
AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough.
* Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the
return value of pj_xml_print() when the supplied buffer is not large
enough.
When a caller calls a FAX number and then hangs up right after the call is
answered then the T.38 re-INVITE automatic reject timer may still be
running after the channel goes away.
* Added session NULL channel checks on the code paths that get executed by
t38_automatic_reject() to prevent a crash when the T.38 re-INVITE
automatic reject timer expires.
Kevin Harwell [Tue, 30 Jun 2015 20:19:31 +0000 (15:19 -0500)]
res_pjsip: Failover when server is not available
Previously Asterisk did not properly failover to the next resolved DNS
address when a endpoint could not be reached. With this patch, and while
using res_pjsip, SIP requests (both in/out of dialog) now attempt to use
the next address in the list of resolved addresses until a proper response
is received or no more addresses are left.
res_sorcery_memory_cache: Execute stale unit test last.
In Jenkins there is currently a sporadic test failure of a
variable number of sorcery memory cache unit tests. I have not
been able to reproduce this on the build agents themselves or
on my development machine.
My working theory is that the stale unit test is causing a
sorcery instance to persist longer than expected, causing subsequent
tests to fail when setting up and initializing the next
sorcery instance.
To see if this is the case this change moves the stale unit test
to execute last so no subsequent unit tests can have issues
initializing their sorcery instance.
Matt Jordan [Sat, 20 Jun 2015 18:54:17 +0000 (13:54 -0500)]
bucket: Add clone/staleness operations for ast_bucket/ast_bucket_file
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.
Matt Jordan [Sat, 4 Jul 2015 15:03:06 +0000 (10:03 -0500)]
sorcery: Add support for object staleness
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.
This change makes it so that when accepting a WebSocket
connection the HTTP response is sent as one packet instead of
fragmented. Browsers don't like it when you send it fragmented.
Matt Jordan [Sat, 27 Jun 2015 23:47:19 +0000 (18:47 -0500)]
Makefile: Remove coverage files on 'make clean'
This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
projects / thirdparty / asterisk.git / log
