* gcc 13 is now catching when a function is declared as returning
an enum but defined as returning an int or vice versa. Fixed
a few in app.h, loader.c, stasis_message.c.
* gcc 13 is also now (incorrectly) complaining of dangling pointers
when assigning a pointer to a local char array to a char *. Had
to change that to an ast_alloca.
Resolves: #155
Adds a new AMI action (CoreShowChannelMap) that takes in a channel name
and provides a list of all channels that are connected to that channel,
following local channel connections as well.
Resolves: #104
UserNote: New AMI action CoreShowChannelMap has been added.
The Caller ID generation routine currently is hardcoded
to always use the system time zone. This makes it possible
to optionally specify any TZ-format time zone.
Resolves: #98
ASTERISK-30330
Commit 09e989f972
categorized the T option as not being compatible
with remote consoles, but they do affect verbose
messages with remote console. This fixes this.
Resolves: #102
If processing an XInclude results in new <xi:include> elements, we
need to run XInclude processing again. This continues until no
replacement occurs or an error is encountered.
There is a separate issue with dynamic strings (ast_str) that will be
addressed separately.
Resolves: #65
The current AST_CEL_LOCAL_OPTIMIZE event is and has been
triggered on a local optimization end to serve as a flag
indicating the event occurred. This change adds a second
AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.
Resolves: #52
UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
ast_waitstream was not called after ast_streamfile,
resulting in "o'clock" being skipped in French.
Additionally, the minute announcements should be
feminine.
Reported-by: Danny Lloyd
Resolves: #41
ASTERISK-30488
The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.
RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.
To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.
Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.
ASTERISK-30407 #close
Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2
The unit test XML output was counting all registered tests as "run"
even when only a subset were actually requested to be run and
the "failures" attribute was missing.
* The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
* The "failures" attribute was added to the "testsuite"
element.
Also added 2 new unit tests that just pass and fail to be
used for CI testing.
Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69
Various changes to ensure that the lexers and parsers can be correctly
generated when REBUILD_PARSERS is enabled.
Some notes:
* Because of the version of flex we are using to generate the lexers
(2.5.35) some post-processing in the Makefile is still required.
* The generated lexers do not contain the problematic C99 check that
was being replaced by the call to sed in the respective Makefiles so
it was removed.
* Since these files are generated, they will include trailing
whitespace in some places. This does not need to be corrected.
Change-Id: Ibbd343606fcf5c0d285b1599e6e8e59f514f2e4e
Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.
If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed. If the interval is set to an
invalid value, it will default to 15 seconds.
A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started. This is so we can test that the touch
variable starts the hook as expected.
ASTERISK-30446
Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661
While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.
This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute. This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.
ASTERISK-30464
Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98
For 'core show channels', the Channel name field is increased
to 64 characters and the Location name field is increased to
32 characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
ASTERISK-30455
Change-Id: Ibec3742ce360ffc93bc56e9984c2a21dabc4d5e1
Change the HTTP status page (located at /httpstatus by default) by:
* Combining the address and port into a single line.
* Changing "SSL" to "TLS"
ASTERISK-30433 #close
Change-Id: Id2ccb1218f00a68424aca2b651647d8b1f549bcb
Fix the following build failure with libressl by using SSL_is_server
which is available since version 2.7.0 and
d7ec516916:
iostream.c: In function 'ast_iostream_close':
iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
559 | if (!stream->ssl->server) {
| ^~
ASTERISK-30107 #close
Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
Change-Id: Iea7f34970297f2fb50285d73462d0174ba7e9587
* Added a new function ast_utf8_replace_invalid_chars() to
utf8.c that copies a string replacing any invalid UTF-8
sequences with the Unicode specified U+FFFD replacement
character. For example: "abc\xffdef" becomes "abc\uFFFDdef".
Any UTF-8 compliant implementation will show that character
as a � character.
* Updated res_pjsip:set_id_from_hdr() to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
* Updated stasis_channels:ast_channel_publish_varset to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
ASTERISK-27830
Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8
This avoids buffer overflow errors when running tests that capture
output from child processes.
This also corrects a copypasta in an off-nominal error message.
Change-Id: Ib482847a3515364f14c7e7a0c0a4213851ddb10d
For most modules that interacted with app_macro, this change is limited
to no longer looking for the current context from the macrocontext when
set. Additionally, the following modules are impacted:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs a re-write
app_queue - can no longer a macro on the called party's channel.
Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected line or
redirection macro options
options - stdexten is deprecated to gosub as the default and only
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
ASTERISK-30304
Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6
-----------------
This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures. The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.
ASTERISK-30391
-----------------
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038
If native HTTP is disabled but HTTPS is enabled and status page enabled
too, Core/HTTP crashes while loading. 'global_http_server' references
to NULL, but the status page tries to dereference it.
The patch adds a check for HTTP is enabled.
ASTERISK-30379 #close
Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249
Currently, if a module declines to load, dlopen is called
to register the module but dlclose never gets called.
Furthermore, loader.c currently doesn't allow dlclose
to ever get called on the module, since it declined to
load and the unload function bails early in this case.
This can be problematic if a module is updated, since the
new module cannot be loaded into memory since we haven't
closed all references to it. To fix this, we now allow
modules to be unloaded, even if they never "loaded" in
Asterisk itself, so that dlclose is called and the module
can be properly cleaned up, allowing the updated module
to be loaded from scratch next time.
ASTERISK-30345 #close
Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5
The if statement here is always false after the for
loop finishes, so variables are never appended.
This removes that to properly append to the end
of the variable list.
ASTERISK-30351 #close
Reported by: Sebastian Gutierrez
Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
ASTERISK-30280
Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a
pbx_exec makes a channel snapshot before executing applications.
This doesn't cause an issue during normal dialplan execution
where pbx_exec is called over and over again in succession.
However, if pbx_exec is called "one off", e.g. using
ast_pbx_exec_application, then a channel snapshot never ends
up getting made after the executed application returns, and
inaccurate snapshot information will linger for a while, causing
"core show channels", etc. to show erroneous info.
This is fixed by manually making a channel snapshot at the end
of ast_pbx_exec_application, since we anticipate that pbx_exec
might not get called again immediately.
ASTERISK-30367 #close
Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086
This removes the ImportVar and SetAMAFlags applications
which have been deprecated since Asterisk 12, but were
never removed previously.
Additionally, it removes remnants of defunct options
that themselves were removed years ago.
ASTERISK-30335 #close
Change-Id: I749520c7b08d4c9d5eebbf640d4fbc81950eda8d
When ast_stream_and_wait returns an error (for example, when attempting
to stream to a channel after hangup) the stream is not closed, and
callers typically do not check the return code. This results in leaking
file descriptors, leading to resource exhaustion.
This change ensures that the stream is closed in case of error.
ASTERISK-30198 #close
Reported-by: Julien Alie
Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803
The XML docs are currently only loaded on
startup with no way to update them during runtime.
This makes it impossible to load modules that
use ACO/Sorcery (which require documentation)
if they are added to the source tree and built while
Asterisk is running (e.g. external modules).
This adds a CLI command to reload the XML docs
during runtime so that documentation can be updated
without a full restart of Asterisk.
ASTERISK-30289 #close
Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07
Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.
ASTERISK-30176
Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404
(cherry picked from commit 81f10e847e)
The Answer application currently waits for up to 500ms
for media, even if users specify a different timeout.
This adds an option to not wait for media on the channel
by doing a raw answer instead. The default 500ms threshold
is also documented.
ASTERISK-30308 #close
Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4
This PR contains two relatively separate changes in channel.c and
res_pjsip_session.c which ensure that topology changes are not ignored
in cases where they should be handled.
For channel.c:
The function ast_channel_request_stream_topology_change only triggers a
stream topology request change indication, if the channel's topology
does not equal the requested topology. However, a channel could be in a
state where it is currently "negotiating" a new topology but hasn't
updated it yet, so the topology request change would be lost. Channels
need to be able to handle such situations internally and stream
topology requests should therefore always be passed on.
In the case of chan_pjsip for example, it queues a session refresh
(re-INVITE) if it is currently in the middle of a transaction or has
pending requests (among other reasons).
Now, ast_channel_request_stream_topology_change always indicates a
stream topology request change even if the requested topology equals the
channel's topology.
For res_pjsip_session.c:
The function resolve_refresh_media_states does not process stream state
changes if the delayed active state differs from the current active
state. I.e. if the currently active stream state has changed between the
time the sip session refresh request was queued and the time it is being
processed, the session refresh is ignored. However, res_pjsip_session
contains logic that ensures that session refreshes are queued and
re-queued correctly if a session refresh is currently not possible. So
this check is not necessary and led to some session refreshes being
lost.
Now, a session refresh is done even if the delayed active state differs
from the current active state and it is checked whether the delayed
pending state differs from the current active - because that means a
refresh is necessary.
Further, the unit test of resolve_refresh_media_states was adapted to
reflect the new behavior. I.e. the changes to delayed pending are
prioritized over the changes to current active because we want to
preserve the original intention of the pending state.
ASTERISK-30184
Change-Id: Icd0703295271089057717006730b555b9a1d4e5a
If multiple codecs are available for the same
resource and the translation costs between
multiple codecs are the same, ties are
currently broken arbitrarily, which means a
lower quality codec would be used. This forces
Asterisk to explicitly use the higher quality
codec, ceteris paribus.
ASTERISK-29455
Change-Id: I4b7297e1baca7aac14fe4a3c7538e18e2dbe9fd6
The ModuleCheck XML documentation falsely
claims that the module's version number is returned.
This has not been the case since 14, since the version
number is not available anymore, but the documentation
was not changed at the time. It is now updated to
reflect this.
ASTERISK-30285 #close
Change-Id: Idde2d1205a11f2623fa1ddab192faa3dc4081e91
write_openssl_error_to_log has been erroneously
using ast_free instead of free, which will
cause a crash when MALLOC_DEBUG is enabled since
the memory was not allocated by Asterisk's memory
manager. This changes it to use the actual free
function directly to avoid this.
ASTERISK-30278 #close
Change-Id: Iac8b6468b718075809c45d8ad16b101af21a474d
race condition: ast_dial_join() may not cancel outgoing call, if
function is called just after called party answer and before
application execution (bit is_running_app not yet set).
This fix adds ast_softhangup() calls in addition to existing
pthread_kill() when is_running_app is not set.
ASTERISK-30258
Change-Id: Idbdd5c15122159661aa8e996a42d5800083131e4
Some logic in say.c for determining if we need
to also add an ampersand for file seperation was faulty,
as non-successful files would increment the count, causing
a leading ampersand to be added improperly.
This is fixed, and a unit test that captures this regression
is also added.
ASTERISK-30248 #close
Change-Id: I02c1d3a11d82fe4ea8b462070cbd1effb5834d2b
Add enum to allow setting optional direction. If set to only one
direction, only feed matching-direction frames to the associated
slin factory.
This prevents mangling the transcoder on non-mixed frames when the
READ and WRITE frames would have otherwise required it. Also
removes the need to mute or discard the un-wanted frames as they
are no longer added in the first place.
res_stasis_snoop is changed to use this addition to set direction
on audiohook based on spy direction.
If no direction is set, the ast_audiohook_init will init this enum
to BOTH which maintains existing functionality.
ASTERISK-30252
Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb
Allows bridging, parking, and dial messages to be globally
ignored for all CDRs such that only a single CDR record
is generated per channel.
This is useful when CDRs should endure for the lifetime of
an entire channel and bridging and dial updates in the
dialplan should not result in multiple CDR records being
created for the call. With the ignore bridging option,
bridging changes have no impact on the channel's CDRs.
With the ignore dial state option, multiple Dials and their
outcomes have no impact on the channel's CDRs. The
last disposition on the channel is preserved in the CDR,
so the actual disposition of the call remains available.
These two options can reduce the amount of "CDR hacks" that
have hitherto been necessary to ensure that CDR was not
"spoiled" by these messages if that was undesired, such as
putting a dummy optimization-disabled local channel between
the caller and the actual call and putting the CDR on the channel
in the middle to ensure that CDR would persist for the entire
call and properly record start, answer, and end times.
Enabling these options is desirable when calls correspond
to the entire lifetime of channels and the CDR should
reflect that.
Current default behavior remains unchanged.
ASTERISK-30091 #close
Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832
The DBGetTree AMI action's ListItem previously
always reported 1, regardless of the count. This
is corrected to report the actual count.
ASTERISK-30245 #close
patches:
gettreecount.diff submitted by Birger Harzenetter (license 5870)
Change-Id: I46d8992710f1b8524426b1255f57d1ef4a4934d4
Adds the n "no answer" option to the Bridge application
so that answer supervision can not automatically
be provided when Bridge is executed.
Additionally, a mechanism (dialplan variable)
is added to prevent bridge targets (typically the
target of a masquerade) from answering the channel
when they enter the bridge.
ASTERISK-30223 #close
Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d
If we find that n_max (currently hard wired to 1) sessions were purged,
schedule the next purge for 1ms into the future rather than 5000ms (as
per current). This way we will purge up to 1000 sessions per second
rather than 1 every 5 seconds.
This mitigates a build-up of sessions should http sessions gets
established faster than 1 per 5 seconds.
Change-Id: I9820d39aa080109df44fe98c1325cafae48d54f5
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Fixed a segfault caused by var_list_from_loc_info() encountering
an empty location info element.
Fixed an issue in ast_strsep() where a value with only whitespace
wasn't being preserved.
Fixed an issue in ast_variable_list_from_quoted_string() where
an empty value was considered a failure.
ASTERISK-30215
Reported by: Dan Cropp
Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a
Adds an AMI event to indicate that a deadlock
has likely started, when Asterisk is compiled
with DETECT_DEADLOCKS enabled. This can make
it easier to perform automated deadlock detection
and take appropriate action (such as doing a core
dump). Unlike the deadlock warnings, the AMI event
is emitted only once per deadlock.
ASTERISK-30161 #close
Change-Id: Ifc6ed3e390f8b4cff7f8077a50e4d7a5b54e42fb
Use const char for char arguments to
pbx_substitute_variables_helper_full_location
that can do so (context and exten).
ASTERISK-30209 #close
Change-Id: I001357177e9c3dca2b2b4eebc5650c1095b3da6f
If "core show channels" is run before startup has completed, it
is possible for bad ao2 refs to occur because the system is not
yet fully initialized. This will lead to an assertion failing.
To prevent this, initialization of CLI builtins is moved to be
later along in the main load sequence. Core CLI commands are
loaded at the same time, but channel-related commands are loaded
later on.
ASTERISK-29846 #close
Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6
Adds additional control options over the transfer
feature functionality to give users more control
in how the transfer feature sounds and works.
First, the "transfer" sound that plays when a transfer is
initiated can now be customized by the user in
features.conf, just as with the other transfer sounds.
Secondly, the user can now specify the transfer extension
in advance by using the TRANSFER_EXTEN variable. If
a valid extension is contained in this variable, the call
will automatically be transferred to this destination.
Otherwise, it will fall back to collecting the extension
from the user as is always done now.
ASTERISK-29899 #close
Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3
Fixes a few coding guideline violations:
* Use of C99 comments
* Opening brace on same line as function prototype
ASTERISK-30163 #close
Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7
* Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
reflect it's purpose.
* Added the config option for 'allow_routing_use' which
sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
* Removed the 'profile' argument.
* Automatically create a profile if it doesn't exist.
* Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.
ASTERISK-30167
Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0
The manager XML documentation documents a "FilterList"
action, but there is no such action. Therefore, this can
lead to confusion when people try to use a documented
action that does not, in fact, exist. This is removed
as the action never did exist in the past, nor would it
be trivial to add since we only store the regex_t
objects, so the filter list can't actually be provided
without storing that separately. Most likely, the
documentation was originally added (around version 10)
in anticipation of something that never happened.
ASTERISK-29917 #close
Change-Id: I846b16fd6f80a91d4ddc5d8a861b522d7c6f8f97
Adjusts some logging levels to be more or less important,
that is more prominent when actual problems occur and less
prominent for less noteworthy things.
ASTERISK-30153 #close
Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b
When ast_func_read2 is used to read a function using
its read function (as opposed to a native ast_str read2
function), the result is copied directly by the function
into the ast_str buffer. As a result, the ast_str length
remains initialized to 0, which is a bug because this is
not the real string length.
This can cascade and have issues elsewhere, such as when
reading substrings of functions that only register read
as opposed to read2 callbacks. In this case, since reading
ast_str_strlen returns 0, the returned substring is empty
as opposed to the actual substring. This has caused
the ast_str family of functions to behave inconsistently
and erroneously, in contrast to the pbx_variables substitution
functions which work correctly.
This fixes this issue by manually updating the ast_str length
when the result is copied directly into the ast_str buffer.
Additionally, an assertion and a unit test that previously
exposed these issues are added, now that the issue is fixed.
ASTERISK-29966 #close
Change-Id: I4e2dba41410f9d4dff61c995d2ca27718248e07f
The global event filtering code was only in one
possible execution path, so not all events were
being properly filtered out if requested. This moves
that into the universal AMI handling code so all
events are properly handled.
Additionally, the CLI listing of disabled events can
also get truncated, so we now print out everything.
ASTERISK-30137 #close
Change-Id: If8c42edcb2abc5158552da7eba2a8ff6b20e1959
Adds the DBGetTree action, which can be used to
retrieve all of the DB keys beginning with a
particular prefix, similar to the capability
provided by the database show CLI command.
ASTERISK-30136 #close
Change-Id: I3be9425e53be71f24303fdd4d2923c14e84337e6
There are several things wrong with analog Caller ID
handling that are fixed by this commit:
callerid.c's Caller ID generation function contains the
logic to use the presentation to properly send the proper
Caller ID. However, currently, DAHDI does not pass any
presentation information to the Caller ID module, which
means that presentation is completely ignored on all calls.
This means that lines could be getting Caller ID information
they aren't supposed to.
Part of the reason this has been obscured is because the
simple switch logic for handling the built in *67 and *82
is completely wrong. Rather than modifying the presentation
for the call accordingly (which is what it's supposed to do),
it simply blanks out the Caller ID or fills it in. This is
wrong, so wrong that it makes a mockery of the specification.
Additionally, it would leave to the "UNAVAILABLE" disposition
being used for Caller ID generation as opposed to the "PRIVATE"
disposition that it should have been using. This is now fixed
to only update the presentation and not modify the number and
name, so that the simple switch *67/*82 work correctly.
Next, sig_analog currently only copies over the name and number,
nothing else, when it is filling in a duplicated caller id
structure. Thus, we also now copy over the presentation
information so that is available for the Caller ID spill.
Additionally, this meant that "valid" was implicitly 0,
and as such presentation would always fail to "Unavailable".
The validity is therefore also copied over so it can be used
by ast_party_id_presentation.
As part of this fix, new API is added so that all the relevant
Caller ID information can be passed in to the Caller ID generation
functions. Parameters that are also completely missing from the
Caller ID spill have also been added, to enhance the compatibility,
correctness, and completeness of the Asterisk Caller ID implementation.
ASTERISK-29991 #close
Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15
Allocate all of the ast_context's character data in the structure's
flexible array member and eliminate the clunky fake_context. This will
simplify future changes to ast_context.
Change-Id: I98357de75d8ac2b3c4c9f201223632e6901021ea
The current documentation is out of date and does not reflect actual
behaviour. This change makes documentation clearer and accurately
reflect the purpose of relevant channel variables.
ASTERISK-30123
Change-Id: I160d0b01fce862477ad55ac1aa708a730473eb6f
* Added ast_variable_list_from_quoted_string()
Parse a quoted string into an ast_variable list.
* Added ast_str_substitute_variables_full2()
Perform variable/function/expression substitution on an ast_str.
* Added ast_strsep_quoted()
Like ast_strsep except you can specify a specific quote character.
Also added unit test.
* Added ast_xml_find_child_element()
Find a direct child element by name.
* Added ast_xml_doc_dump_memory()
Dump the specified document to a buffer
* ast_datastore_free() now checks for a NULL datastore
before attempting to destroy it.
Change-Id: I5dcefed2f5f93a109e8b489e18d80d42e45244ec
Currently, if using the CLI to delete a DB entry,
"Database entry removed" is always returned,
regardless of whether or not the entry actually
existed in the first place. This meant that users
were never told if entries did not exist.
The same issue occurs if trying to delete a DB key
using AMI.
To address this, new API is added that is more stringent
in deleting values from AstDB, which will not return
success if the value did not exist in the first place,
and will print out specific error details if available.
ASTERISK-30001 #close
Change-Id: Ic84e3eddcd66c7a6ed7fea91cdfd402568378b18
A corner case exists in CLI parsing where if
a CLI user in a remote console ends with
a backslash and then invokes command completion
(using TAB or ?), then the console will freeze
forever until a SIGQUIT signal is sent to the
process, due to getting blocked forever
reading the command completion. CTRL+C
and other key combinations have no impact on
the CLI session.
This occurs because, in such cases, the CLI
process is waiting for AST_CLI_COMPLETE_EOF
to appear in the buffer from the main process,
but instead the main process is confused by
the funny syntax and thus prints out the CLI help.
As a result, the CLI process is stuck on the
read call, waiting for the completion that
will never come.
This prevents blocking forever by checking
if the data from the main process starts with
"Usage:". If it does, that means that CLI help
was sent instead of the tab complete vector,
and thus the CLI should bail out and not wait
any longer.
ASTERISK-29822 #close
Change-Id: I9810ac59304fec162da701653c9c834f0ec8f670
If the caller has hung up, break out of the play loop so we don't try
to play remaining files and fail to do so.
ASTERISK-30075 #close
Change-Id: I55e85be28ee90b48c0fe4ce20ac136a7dbb49f14
Finding an application and executing it if found is
a common task throughout Asterisk. This adds a helper
function around pbx_exec to do this, to eliminate
redundant code and make it easier for modules to
substitute variables and execute applications by name.
ASTERISK-30061 #close
Change-Id: Ifee4d2825df7545fb515d763d393065675140c84
A previous review fixing ASTERISK_22246 and ASTERISK_26582
got a couple of the options mixed up as to whether or not
they are compatible with the remote console. This fixes
those to the best of my knowledge.
ASTERISK-30097 #close
Change-Id: Id54166991aa79f04fb02699cc499bedda854253b
In function ast_say_date_with_format_de(), take special
care when the hour is one o'clock. In this case, the
German number "eins" must be inflected to its neutrum form,
"ein". This is achieved by playing "digits/1N" instead of
"digits/1". Fixes both 12- and 24-hour formats.
ASTERISK-30092
Change-Id: Ica9b80125c0b317e378d89c1ea786816e2635510
If tab completion using ast_module_helper is attempted
during startup, deadlock will ensue because the CLI
will attempt to lock the module list while it is already
locked by the loader. This causes deadlock because when
the loader tries to acquire the CLI lock, they are blocked
on each other.
Waiting for startup to complete is not feasible because
the CLI lock is acquired while waiting, so deadlock will
ensure regardless of whether or not a lock on the module
list is attempted.
To prevent deadlock, we immediately abort if tab completion
is attempted on the module list before Asterisk is fully
booted.
ASTERISK-30039 #close
Change-Id: Idd468906c512bb196631e366a8f597a0e2e9271d
This change exposes the channel driver's unique id (i.e. the Call-ID
for chan_sip/chan_pjsip based channels) to ARI channel resources
as `protocol_id`.
ASTERISK-30027
Reported by: Moritz Fain
Tested by: Moritz Fain
Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87
Most issues were in stringfields and had to do with comparing
a pointer to an constant/interned string with NULL. Since the
string was a constant, a pointer to it could never be NULL so
the comparison was always "true". gcc now complains about that.
There were also a few issues where determining if there was
enough space for a memcpy or s(n)printf which were fixed
by defining some of the involved variables as "volatile".
There were also a few other miscellaneous fixes.
ASTERISK-30044
Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570
GCC 12 caught an issue in state_id_by_topic where we were
checking a pointer for NULL instead of the contents of
the pointer for '\0'.
ASTERISK-30044
Change-Id: Ia0b04d4fff45c92acb7f07132a33622fa341148e
When a new unreal (local) channel is created, a second (;2) channel is
created as a counterpart which clones the topology of the first
channel. This creates issues when an outgoing stream is sendonly or
recvonly as the stream state of the inbound channel will be the same
as the stream state of the outbound channel.
Now the stream state is flipped for the streams of the 2nd channel in
ast_unreal_new_channels if the outgoing stream topology is recvonly or
sendonly.
ASTERISK-29655
Reported by: Michael Auracher
ASTERISK-29638
Reported by: Michael Auracher
Change-Id: I0cea29635bb20b7bf7fd0fb95498cd44dab98fbf
Some command line options to Asterisk only apply when Asterisk
is started and cannot be used with remote console mode. If a
user tries to use any of these, they are currently simply
silently ignored.
This prints out a warning if incompatible options are used,
informing users that an option used cannot be used with remote
console mode. Additionally, some clarifications are added to
the help text and man page.
ASTERISK-22246
ASTERISK-26582
Change-Id: I980a5380ef2c19e8ea348596396d5382893c4337
This adds the EVAL_EXTEN function, which may be used to retrieve
the variable-substituted data at any extension.
ASTERISK-29486
Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432
Currently, if a user uses an application like ControlPlayback
to try to rewind a file past the beginning, this can throw
warnings when the file format (e.g. PCM) tries to seek to
a negative offset.
Instead of letting file formats try (and fail) to seek a
negative offset, we instead now catch this in the rewind
function to ensure that we never seek an offset less than 0.
This prevents legitimate user actions from triggering warnings
from any particular file formats.
ASTERISK-29943 #close
Change-Id: Ia53f2623f57898f4b8e5c894b968b01e95426967
Adds the dialplan eval function commands to evaluate a dialplan
function from the CLI. The return value and function result are
printed out and can be used for testing or debugging.
ASTERISK-29820 #close
Change-Id: I833e97ea54c49336aca145330a2adeebfad05209
Adds version information for applications, functions,
and manager events/actions.
This is not completely exhaustive by any means but
covers most new things added that have release
versioning information in the issue tracker.
ASTERISK-29940 #close
Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131
On a write error to an AMI session a flag was set to
indicate that the write error had occurred, with the
expected result being that the session be terminated.
This was not actually happening and instead writing
would continue to be attempted.
This change adds a check for the write error and causes
the session to actually terminate.
ASTERISK-29948
Change-Id: Icaf5d413d4c0d5dc78292a17287fecc8720a31a5
The PBX core uses the stack when it comes to includes, which
means that a context can only contain strictly fewer than
AST_PBX_MAX_STACK includes. If this is exceeded, then warnings
will be emitted for each number of includes beyond this if
searching for an extension in the including context, and if
the extension's inclusion is beyond the stack size, it will
simply not be found.
To address this, we now check if there are too many includes
in a context when the dialplan is reloaded so that if there
is an issue, the user is aware of at "compile time" as opposed
to "run time" only. Secondly, more details are printed out
when this message is encountered so it's clear what has happened.
ASTERISK-26719
Change-Id: Ia3700452e75a7af3391b3e82ee69f06a669f8958
Treat time_t's as entirely unique and use the POSIX API's for
converting to/from strings.
Lastly, a 64-bit integer formats as 20 digits at most in base10.
Don't need to have any 100 byte buffers to hold that.
ASTERISK-29674 #close
Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56
Passing 0 as the last argument to strtoimax() or strtoumax() causes
octal and hexadecimal to be accepted which was not originally
intended. So we now force to only accept decimal.
ASTERISK-29950 #close
Change-Id: I93baf0f273441e8280354630a463df263a8c0edd
MUSL defines BUFSIZ as 1024 which is not reasonable for log messages.
More broadly, BUFSIZ is the amount of buffering stdio.h does, which
is arbitrary and largely orthogonal to what logging should accept
as the maximum message size.
ASTERISK-29928
Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Iaa49fbbab029c64ae3d95e4b18270e0442cce170
BackGround and WaitExten both accept options that are not
currently documented. This adds documentation for these
options to the xml documentation for each application.
ASTERISK-29967 #close
Change-Id: If812a9f1ccbba3e4d427a0e7a6dea923c2f905f7
ASTERISK_29853 added the ability to selectively disable
AMI events on a global basis, but the logic for this uses
strstr which means that events with names which are the prefix
of another event, if disabled, could disable those events as
well.
Instead, we account for this possibility to prevent this
undesired behavior from occuring.
ASTERISK_29853
Change-Id: Icccd1872602889806740971e4adf932f92466959
Added functions to open, close, and apply XML Stylesheets
to XML documents. Although the presence of libxslt was already
being checked by configure, it was only happening if xmldoc was
enabled. Now it's checked regardless.
Added ability to parse a string consisting of comma separated
name/value pairs into an ast_variable list. The reverse of
ast_variable_list_join().
Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e
Added:
Replace a variable in a list:
int ast_variable_list_replace_variable(struct ast_variable **head,
struct ast_variable *old, struct ast_variable *new);
Added test as well.
Create a "name=value" string from a variable list:
'name1="val1",name2="val2"', etc.
struct ast_str *ast_variable_list_join(
const struct ast_variable *head, const char *item_separator,
const char *name_value_separator, const char *quote_char,
struct ast_str **str);
Added test as well.
Allow the name of an XML element to be changed.
void ast_xml_set_name(struct ast_xml_node *node, const char *name);
Change-Id: I330a5f63dc0c218e0d8dfc0745948d2812141ccb
Adds the lastcontext and lastexten channel fields to allow users
to access previous dialplan execution locations.
ASTERISK-29840 #close
Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307
Although there are 10 debugs levels, over time,
many current debug calls have come to use
inappropriately low debug levels. In particular,
a select few debug calls (currently all debug 1)
can result in thousands of debug messages per minute
for a single call.
This can adds a lot of noise to core debug
which dilutes the value in having different
debug levels in the first place, as these
log messages are from the core internals are
are better suited for higher debug levels.
Some debugs levels are thus adjusted so that
debug level 1 is not inappropriately overloaded
with these extremely high-volume and general
debug messages.
ASTERISK-29897 #close
Change-Id: I55a71598993552d3d64a401a35ee99474770d4b4
Adds two pieces of information to the core show settings command
which are useful in the context of getting backtraces.
The first is to display whether or not Asterisk would generate
a core dump if it were to crash.
The second is to show the current running directory of Asterisk.
ASTERISK-29866 #close
Change-Id: Ic42c0a9ecc233381aad274d86c62808d1ebb4d83
The configObject tag contains a default attribute which
allows the default value to be specified, if applicable.
This allows for the default value to show up specially on
the wiki in a way that is clear to users.
There are a couple places in the tree where default values
are included in the description as opposed to as attributes,
which means these can't be parsed specially for the wiki.
These are changed to use the attribute instead of being
included in the text description.
ASTERISK-29898 #close
Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755
The disabledevents setting has been added to the general section
in manager.conf, which allows users to specify events that
should be globally disabled and not sent to any AMI listeners.
This allows for processing of these AMI events to end sooner and,
for frequent AMI events such as Newexten which users may not have
any need for, allows them to not be processed. Additionally, it also
cleans up core debug as previously when debug was 3 or higher,
the debug was constantly spammed by "Analyzing AMI event" messages
along with a complete dump of the event contents (often for Newexten).
ASTERISK-29853 #close
Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205
When tps_shutdown is called as part of the cleanup process there is a
chance that one of the taskprocessors that references the
tps_singletons object is still running. The change is to allow for
tps_shutdown to check tps_singleton's container count and give the
running taskprocessors a chance to finish. If after
AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT (10) seconds there are still
container references we shutdown anyway as this is most likely a bug
due to a taskprocessor not being unreferenced.
ASTERISK-29365
Change-Id: Ia932fc003d316389b9c4fd15ad6594458c9727f1
This code was needlessly complex and would fail to properly delimit
the response message if LOW_MEMORY was defined.
Change-Id: Iae50bf09ef4bc34f9dc4b49435daa76f8b2c5b6e
The XML Manager Event Interface (amxml) now generates attribute names
that are compliant with the XML 1.1 specification. Previously, an
attribute name that started with a digit would be rendered as-is, even
though attribute names must not begin with a digit. We now prefix
attribute names that start with a digit with an underscore ('_') to
prevent XML validation failures.
This is not backwards compatible but my assumption is that compliant
XML parsers would already have been complaining about this.
ASTERISK-29886 #close
Change-Id: Icfaa56a131a082d803e9b7db5093806d455a0523
Adds a new option, defaultenabled, to the CDR core to
control whether or not CDR is enabled on a newly created
channel. This allows CDR to be disabled by default on
new channels and require the user to explicitly enable
CDR if desired. Existing behavior remains unchanged.
ASTERISK-29808 #close
Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1
sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an
executing call-back. This is done by adding a new variable 'rescheduled'
to the struct sched which is set in ast_sched_runq and checked in
ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for
now deprecated ast_sched_del which returns a new possible value -2
if called on an executing call-back with rescheduled set. ast_sched_del
is modified to call ast_sched_del_nonrunning to maintain existing code.
AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it
will not throw a warning or invoke refcall.
test_sched: Add a new unit test sched_test_freebird that will check the
reference count in the resolved scenario.
ASTERISK-29698
Change-Id: Icfb16b3acbc29cf5b4cef74183f7531caaefe21d
Implement the ast_get_tid() function for NetBSD system. NetBSD supports
getting the TID via _lwp_self().
ASTERISK-29850
Change-Id: If57fd3f9ea15ef5d010bfbdcbbbae9b379f72f8c
Enable the Linux rdtsc implementation on NetBSD as well. The assembly
works correctly there.
ASTERISK-29851
Change-Id: I460ad9b4d971913420ecb84186f5ba5ab03f6f37
A regression was introduced in ASTERISK~29531 that caused 'say'
functions to fail with file lists that would previously have
succeeded. This caused affected channels to hang up where previously
they would have continued.
We now explicitly check for the empty string to restore the previous
behavior.
ASTERISK-29859 #close
Change-Id: Ia2e5769868e2792313c2d7c07996efe009c6f8d5
Every config variable in the directories
section of asterisk.conf currently has a
counterpart built-in variable containing
the value of the config option, except
for the last one, astsbindir, which should
have an ASTSBINDIR variable.
However, the actual corresponding ASTSBINDIR
variable is missing in pbx_variables.c.
This adds the missing variable so that all
the config options have their corresponding
variable.
ASTERISK-29847 #close
Change-Id: I36006faf471825b36ebc8aa5e87a3bcb38d446fc
gethostbyname() and gethostbyname_r() are deprecated in favor of
getaddrinfo() which we use in the ast_sockaddr family of functions.
ASTERISK-29819 #close
Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7
Adds the macro DTMF_MATRIX_SIZE to replace
the magic number 4 sprinkled throughout
dsp.c.
ASTERISK-29815 #close
Change-Id: Ie3bddb92c6b16204ece0f758009e9490eb33b9ba
Adds a command to the CLI to unload and then
load a module. This makes it easier to perform
these operations which are often done
subsequently to load a new version of a module.
"module reload" already refers to reloading of
configuration, so the name "refresh" is chosen
instead.
ASTERISK-29807 #close
Change-Id: I595f6f11774a0de2565a1fba38da22309ce93a2c
Adds missing documentation for some channel,
bridge, and queue events.
ASTERISK-24427
ASTERISK-29515
Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64
The current TCP client connect code, blocks and does not handle EINTR
error case.
This patch makes the client socket non-blocking while connecting,
ensures a connect does not immediately fail due to EINTR "errors",
and adds a connect timeout option.
The original client start call sets the new timeout option to
"infinite", thus making sure old, orginal behavior is retained.
ASTERISK-29746 #close
Change-Id: I907571843a83e43c0742b95a64785f4411f02671
Adds tech-agnostic support for SF signaling
by adding SF sender and receiver applications
as well as Dial integration.
ASTERISK-29802 #close
Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad
SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal,
and SayPhonetic all claim to allow DTMF interruption if the
SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we
are failing to break out of a given 'say' application if DTMF actually
occurs.
ASTERISK-29816 #close
Change-Id: I6a96e0130560831d2cb45164919862b9bcb6287e
It's not safe to keep the channel locked while locking
the peer Local channel, as it can result in a deadlock.
This change unlocks it during this time but keeps the
bridge locked to ensure nothing changes about the bridge.
ASTERISK-29821
Change-Id: Ib68eb7037e5a479bcc2aceee77337cdde1fbdde6
The variable cp4 in a variable substitution function
can potentially be used without being initialized
currently. This causes Asterisk to no longer compile.
This initializes cp4 to NULL to make the compiler
happy.
ASTERISK-29803 #close
Change-Id: I392579cbb76db2795d5820c9427cf55fbcee9e72
Previously, it was only possible to have one HTTP server in Asterisk.
With this patch it is now possible to have multiple HTTP servers
listening on different addresses.
Note, this behavior has only been made available through an API call
from within the TEST_FRAMEWORK. Specifically, this feature has been
added in order to allow unit test to create/start and stop servers,
if one has not been enabled through configuration.
Change-Id: Ic5fb5f11e62c019a1c51310f4667b32a4dae52f5
Currently, Asterisk doesn't throw warnings if options
are passed into applications that don't accept them.
This can confuse users if they're unaware that they
are doing something wrong.
This adds an additional check to parse_options so that
a warning is thrown anytime an option is parsed that
doesn't exist in the parsing application, so that users
are notified of the invalid usage.
ASTERISK-29801 #close
Change-Id: Id029274a57135caca193c913307a63fd75e24679
Currently MSet can only parse a maximum of 24 variables.
If more variables are provided to MSet, the 24th variable
will simply contain the remainder of the string and the
remaining variables thereafter will never get set.
This increases the number of variables that can be parsed
in one go from 24 to 99. Additionally, documentation is added
since this limitation is currently undocumented and is
confusing to users who encounter this limitation.
ASTERISK-29766 #close
Change-Id: I3fe35b462dedec0a452fd9ea7f92c920a3939f16
Currently, variable substitution involving dialplan
extensions is quite clunky since it entails obtaining
the current dialplan location, backing it up, storing
the desired variables for substitution on the channel,
performing substitution, then restoring the original
location.
In addition to being clunky, things could also go wrong
if an async goto were to occur and change the dialplan
location during a substitution.
Fundamentally, there's no reason it needs to be done this
way, so new API is added to allow for directly passing in
the dialplan location for the purposes of variable
substitution so we don't need to mess with the channel
information anymore. Existing API is not changed.
ASTERISK-29745 #close
Change-Id: I23273bf27fa0efb64a606eebf9aa8e2f41a065e4
Adds tech-agnostic support for MF signaling by adding
MF sender and receiver applications as well as Dial
integration.
ASTERISK-29496-mf #do-not-close
Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4
We know that passing a NULL or empty argument to
ast_channel_get_by_name() will never result in a matching channel and
will always result in an error being emitted, so just short-circuit
out in that case.
ASTERISK-28219 #close
Change-Id: I88eadc748e9c6996fc17467b0a05881bbfd00bce
Since Doxygen 1.8.16, a special comment block is required. Otherwise
(pure C comment), the group command is ignored. Additionally, several
unbalanced group commands were fixed.
ASTERISK-29732
Change-Id: I4687857b9d56e6f44fd440b73af156691660202e
A backend's implementation of the realtime 'require' function may call
va_arg() and then fail, leaving the va_list in an undefined
state. Pass a copy of the va_list instead.
ASTERISK-29771 #close
Change-Id: I555565a72af84e96d49f62fe8cb66ba5a78461f4
Refactors generic functions used for email generation
into utils.c so that they can be used by multiple
modules, including app_voicemail and app_minivm,
to avoid code duplication.
ASTERISK-29715 #close
Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee
In the AO2_ALLOC_OPT_LOCK_NOLOCK case the referenced obj
structure is freed, but is then referenced later if ref_log is
enabled. The change is to store the obj->priv_data.options value
locally and reference it instead of the value from the freed obj
ASTERISK-29730
Change-Id: I60cc5dc1f5a4330e7ad56976fc38a42de0ab6072
Local channels are made up of two pairs - the 1 and 2
sides. When a frame goes in one side, it comes out the
other. Back and forth. When both halves are in a
bridge this creates an infinite loop of frames.
This change makes it so that bridging no longer
allows both of these sides to exist in the same
bridge.
ASTERISK-29748
Change-Id: I29928b6de87cd9be996a77daccefd7c360fef651
Fixes four misuses of the parameter 'name'. Additionally, for
consistency and to avoid such an issue in future, those few other
places, which used '\file name', were changed just to '\file'. Then,
Doxygen uses the name of the current file.
ASTERISK-29733
Change-Id: I0c18b4c863c6988b138c77448057349a9ee7052d
Currently, when the t option is specified with no arguments,
the # character is still treated as a terminator, even though
no character should be treated as a terminator.
This is because a previous regression fix was modified to
remove the use of NULL as a default altogether. However,
NULL and an empty string actually refer to different
arrangements and should be treated differently. NULL is the
default terminator (#), while an empty string removes the
terminator altogether. This is the behavior being used by
the rest of the core.
Additionally, since S_OR catches empty strings as well as
NULL (not intended), this is changed to a ternary operator
instead, which fixes the behavior.
ASTERISK-29705 #close
Change-Id: I9b6b72196dd04f5b1e0ab5aa1b0adf627725e086
When reloading dialplan, hints created dynamically would lose any dash
characters. Now we ignore those dashes if we are dealing with a hint
during a reload.
ASTERISK-28040 #close
Change-Id: I95e48f5a268efa3c6840ab69798525d3dce91636
* Initialize some variables that are never used anyway.
* Use valid pointers instead of integers cast to void pointers when
calling pthread_setspecific().
ASTERISK-29711 #close
ASTERISK-29713 #close
Change-Id: I8728cd6f2f4b28e0e48113c5da450b768c2a6683
Add a function to check if there is an exact match a one string between
delimiters in another string.
Add a function that will create an ast_json object out of a list of
Asterisk variables. An excludes string can also optionally be passed
in.
Also, add a macro to make it easier to get object integers.
Change-Id: I5f34f18e102126aef3997f19a553a266d70d6226
Some ast_stun_request users do not provide a destination address when
sending to a connection-mode socket.
ASTERISK-29691
Change-Id: Idd9114c3380216ba48abfc3c68619e79ad37defc
The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today
ASTERISK-29637
Change-Id: I1fb1cef0ce3c18d87b1fc94ea309d13bc344af02
The MessageSend AMI action has been updated to allow the Destination
and the To addresses to be provided separately. This brings the
MessageSend manager command in line with the capabilities of the
MessageSend dialplan application.
ASTERISK-29663 #close
Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c
Adds the ability for users to log to custom log levels
by providing custom log level names in logger.conf. Also
adds a logger show levels CLI command.
ASTERISK-29529
Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702
Up until now, all of the logic used to translate
arguments to the Say applications has been
directly coupled to playback, preventing other
modules from using this logic.
This refactors code in say.c and adds a SAYFILES
function that can be used to retrieve the file
names that would be played. These can then be
used in other applications or for other purposes.
Additionally, a SayMoney application and a SayOrdinal
application are added. Both SayOrdinal and SayNumber
are also expanded to support integers greater than
one billion.
ASTERISK-29531
Change-Id: If9718c89353b8e153d84add3cc4637b79585db19
dsp.c contains arbitrary tone detection functionality
which is currently only used for fax tone recognition.
This change makes this functionality publicly
accessible so that other modules can take advantage
of this.
Additionally, a WaitForTone and TONE_DETECT app and
function are included to allow users to do their
own tone detection operations in the dialplan.
ASTERISK-29546
Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26
ncurses 6.1 introduced an extended number format for terminfo files
which the terminfo parsing in Asterisk is not able to parse. This
results in some TERM values that do support color (screen-256color on
Ubuntu 20.04 for example) to not get a color console.
ASTERISK-29630 #close
Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3
IPv6 nameserver addresses are stored in different part of the
__res_state structure, so look there if we appear to have support for
it.
ASTERISK-28004 #close
Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d
There are 3 separate changes here but they are all closely related:
* Only try to set matchfield attributes on 'field' nodes
* We need to adjust how we treat the category pointer based on the
value of the category_match, to avoid memory corruption. We now
generate a regex-like string when match types other than
ACO_WHITELIST and ACO_BLACKLIST are used.
* Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
ACO_BLACKLIST_EXACT since we only have one category we need to
ignore, not two.
ASTERISK-29614 #close
Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e
Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.
ASTERISK-18454 #close
Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b
This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.
The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.
ASTERISK-29508 #close
Change-Id: I7955a046293f913ba121bbd82153b04439e3465f
The attended transfer feature will emit a warning if the user
cancels the transfer or the attended transfer doesn't complete
for any reason. Changes the warning to a verbose message,
since nothing is actually wrong here.
ASTERISK-29612 #close
Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d
When playing a remote sound file, which is not in cache, first we need
to download it with ast_bucket_file_retrieve.
This can take a while if the remote host is slow. The current CURL
timeout is 180secs, so in extreme situations, it can take 3 minutes to
return.
Because ast_media_cache_retrieve has a lock on all function, while we
are waiting for the delayed download, Asterisk is not able to play any
more files, even the files already cached locally.
ASTERISK-29544 #close
Change-Id: I8d4142b463ae4a1d4c41bff2bf63324821567408
With Asterisk 1.6.0, in the main parser for the configuration file
extensions.conf, the separator was changed from vertical bar to comma.
However, the first separator was not changed in aelparse; it still had
to be a vertical bar, and no comma was allowed.
Additionally, this change allows the vertical bar for the first and
last parameter again, even in the main parser, because the vertical bar
was still accepted for the other parameters.
ASTERISK-29540
Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81
If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps.
This makes the audio stream not-playable at the receiver side.
Linphone isn't able to play such an audio - lots of disruptions are heard.
Also I had complains of bad audio from users which use other types of phones.
After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c).
Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c).
However, this flag is never set in Asterisk-12 and newer.
Previously it has been set (see Asterisk-11).
ASTERISK-29526 #close
Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d
Use cURL's URL parsing API, falling back to the urlparser library, to
parse playback URLs in order to find their file extensions.
For backwards compatibility, we first look at the full URL, then at
any Content-Type header, and finally at just the path portion of the
URL.
ASTERISK-27871 #close
Change-Id: I16d0682f6d794be96539261b3e48f237909139cb
This appears to just have been a copy/paste error from 6258bbe7. Fix
suggested by Ross Beer in ASTERISK~29166.
Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37
If the system time has stepped backwards because of a time
adjustment between the time a frame is timestamped and the
time we check the timestamps in abstract_jb:hook_event_cb(),
we get a negative interval, but we don't check for that there.
abstract_jb:hook_event_cb() then calls
fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed)
and the first thing that does is assert(interval >= 0).
There are several issues with this...
* abstract_jb:hook_event_cb() saves the interval in a variable
named "now" which is confusing in itself.
* "now" is defined as an unsigned int which converts the negative
value returned from ast_tvdiff_ms() to a large positive value.
* fixed_jb_get()'s parameter is defined as a signed int so the
interval gets converted back to a negative value.
* fixed_jb_get()'s assert is NOT an ast_assert but a direct define
that points to the system assert() so it triggers even in
production mode.
So...
* hook_event_cb()'s "now" was renamed to "relative_frame_start" and
changed to an int64_t.
* hook_event_cb() now checks for a negative value right after
retrieving both the current and framedata timestamps and just
returns the frame if the difference is negative.
* fixed_jb_get()'s local define of ASSERT() was changed to call
ast_assert() instead of the system assert().
ASTERISK-29480
Reported by: Dan Cropp
Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9
When using the Busy() and Congestion() applications the
function ast_safe_sleep is used by wait_for_hangup to safely
wait on the channel. This function may send silence if Asterisk
is configured to do so using the transmit_silence option.
In a scenario where an answered channel dials a Local channel
either directly or through call forwarding and the Busy()
or Congestion() dialplan applications were executed with the
transmit_silence option enabled the busy or congestion
tone would not be heard.
This is because inband generation of tones (such as busy
and congestion) is stopped when other audio is sent to
the channel they are being played to. In the given
scenario the transmit_silence option would result in
silence being sent to the channel, thus stopping the
inband generation.
This change adds a variant of ast_safe_sleep which can be
used when silence should not be played to the channel. The
wait_for_hangup function has been updated to use this
resulting in the tones being generated as expected.
ASTERISK-29485
Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133
Previously, SayNumber always emitted a warning if the caller hung up
during execution. Usually this isn't correct, so check if the channel
hung up and, if so, don't emit a warning.
ASTERISK-29475
Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594
Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.
This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.
ASTERISK-29380
Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81
AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
where it should be ignored. Adding this to the switch ensures a
warning isn't thrown on RFC2833 flash events, since nothing's amiss.
ASTERISK-29372
Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc
STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.
The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c
Enhancements:
* The MessageSend dialplan application now takes an optional
third argument that can set the message's "To" field on
outgoing messages. It's an alternative to using the
MESSAGE(to) dialplan function.
NOTE: No channel driver currently implements this field. A
follow-on commit for res_pjsip_messaging will implement it for
the chan_pjsip channel driver.
* To prevent confusion with the first argument, currently named
"to", it's been renamed to "destination". Its function,
creating the request URI, hasn't changed.
* The documentation for MessageSend was updated to be
more clear about the parameters and how they interact
the MESSAGE() dialplan function.
* With the rename of MessageSend's first parameter, and the fact
that message.c references <info> elements in chan_sip.c,
res_pjsip_messaging.c and res_xmpp, they each needed
documentation updates to use MessageDestinationInfo instead of
MessageToInfo.
* appdocsxml.dtd was updated to include a missing element
declaration for "dataType". This was showing up as an error
in Eclipse's dtd editor.
* Despite the changes in this commit, there should be
no impact to current users of MessageSend.
Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a
When a stream topology is provided to chan_local when dialing
it filters the audio formats down. This operation did not skip
streams which were removed (that have no formats) resulting in
calling being aborted.
This change causes such streams to be skipped.
ASTERISK-29407
Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6
Up/down sampling changes the number of samples produced by a translation.
This must be taken into account when checking the codec's buffer size.
ASTERISK-29328
Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055
There is a possibility, when bridge_channel_write_frame() is
called, that the bridge_channel->chan will be NULL. The first
thing bridge_channel_write_frame() does though is call
ast_channel_is_multistream() which had no check for a NULL
channel and therefore caused a segfault. Since it's still
possible for bridge_channel_write_frame() to write the frame to
the other channels in the bridge, we don't want to bail before we
call ast_channel_is_multistream() but we can just skip the
multi-channel stuff. So...
bridge_channel_write_frame() only calls ast_channel_is_multistream()
if bridge_channel->chan is not NULL.
As a safety measure, ast_channel_is_multistream() now returns
false if the supplied channel is NULL.
ASTERISK-29379
Reported-by: Vyrva Igor
Reported-by: Ross Beer
Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce
Using the information from the MODULEINFO XML we can
now output useful information at the end of module
loading for deprecated modules. This includes the
version it was deprecated in, the version it will be
removed in, and the replacement if available.
ASTERISK-29339
Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a
Added a TIME_UNIT enumeration, and a function that converts a
string to one of the enumerated values. Also, added functions
that create and initialize a timeval object using a specified
value, and unit type.
Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392
The 'core' console (ie: asterisk -c) does read logger.conf and does
use the dateformat= option.
Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
and uses a hard coded dateformat option for printing received verbose messages:
main/logger.c: static char dateformat[256] = "%b %e %T"
This change will load logger.conf for each remote console session and
use the dateformat= option to set the per-line timestamp for verbose messages
Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
ASTERISK-25358: #close
Reported-by: Igor Liferenko
When using the ast_unreal_lock_all function no channel
locks can be held before calling it.
This change unlocks the channel that indicate was
called on before doing so and then relocks it afterwards.
ASTERISK-29035
Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9
There exists an inconsistency with framehook usage
such that it is only on reads that the frame should
be freed, not on writes as well.
ASTERISK-29071
Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472
Some sorcery objects actually contain dynamic content
that can change despite the underlying configuration
itself not changing. A good example of this is the
res_pjsip_endpoint_identifier_ip module which allows
specifying hostnames. While the configuration may not
change between reloads the DNS information of the
hostnames can.
This change adds the ability for a sorcery object to be
marked as having dynamic contents which is then taken
into account when reloading by the sorcery file based
config module. If there is an object with dynamic content
then a reload will be forced while if there are none
then the existing behavior of not reloading occurs.
ASTERISK-29321
Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1
A frame suppression API exists as part of channels
which allows audio frames to or from a channel to
be dropped. The MuteAudio AMI action uses this
API to perform its job.
This API uses a framehook to intercept flowing
audio and drop it when appropriate. It is the
responsibility of the framehook to free the
frame it is given if it changes the frame. The
suppression API failed to do this resulting in
a leak of audio frames.
This change adds the freeing of these frames.
ASTERISK-29071
Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b
Implemented the english way of saying the year in ast_say_date_with_format_nl.
Currently the numbers are spoken correctly until 2020 and stopped working
this year.
ASTERISK-29297 #close
Reported-by: Jacek Konieczny
Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4
Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.
This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.
Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34
After some changes to streams and topologies, receiving fax through
local channels stopped working. This change adds a stream topology with
a stream of type IMAGE to the local channel pair and allows fax to be
received.
ASTERISK-29035 #close
Change-Id: Id103cc5c9295295d8e68d5628e76220f8f17e9fb
When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received
This allows applications to perform actions based on the failure
reason.
ASTERISK-29252 #close
Reported-by: Dan Cropp
Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4
Rename check_manager_enabled() and check_webmanager_enabled() to begin
with ast_ so that the symbols are automatically exported by the
linker.
ASTERISK~29184
Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9
This was dead code, test code introduced with Asterisk 13. This was
found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
partly fixes, not closes those two issues.
Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940
The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.
https://wiki.asterisk.org/wiki/x/Xc5uAg
However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.
For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.
ASTERISK-28883 #close
Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f
Scope tracing allows you to not specify a format string or variable,
in which case it just prints the indent, file, function, and line
number. The trace output automatically adds a newline to the end
in this case. If you also have debugging turned on for the module,
a debug message is also printed but the standard log functionality
which prints it doesn't add the newline so you have messages
that don't break correctly.
* format_log_message_ap(), which is the common log
message formatter for all channels, now adds a
newline to the end of format strings that don't
already have a newline.
ASTERISK-29209
Reported by: Alexander Traud
Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da
George Joseph
Ben Ford
Mike Bradeen
Naveen Albert
InterLinked1
Sean Bright
Sean Bright
Boris P. Korzun
Fabrice Fontaine
George Joseph
Peter Fern
Michael Kuron
Maximilian Fridrich
Frederic LE FOLL
Philip Prindeville
Jaco Kroon
Trevor Peirce
Christof Efkemann
Moritz Fain
Joshua C. Colp
Sean Bright
Michał Górny
Kevin Harwell
Alexander Traud
Josh Soref
Sebastien Duthil
Shloime Rosenblum
Andre Barbosa
under
Naveen Albert
Jean Aunis
Mark Murawski
Nico Kooijman
Dan Cropp
Ivan Poddubnyi