In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.
Resolves: #71
UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
Adds last locked and unlocked timestamps as well as a
counter for the number of times the lock has been
attempted (vs locked/unlocked) to debug output printed
using the DEBUG_THREADS option.
Resolves: #110
If we don't set this to -1 if the structure can be potentially re-used
later then it's possible that we'll issue a close() on an unrelated file
descriptor, breaking asterisk in other interesting ways.
I believe this to be an unlikely scenario, but it costs nothing to be
safe.
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
* 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
Maximilian Fridrich
Joshua C. Colp
Mike Bradeen
Jaco Kroon
George Joseph
Ben Ford
Naveen Albert
InterLinked1
Sean Bright
Sean Bright
Boris P. Korzun
Fabrice Fontaine
George Joseph
Peter Fern
Michael Kuron
Frederic LE FOLL