Send "RELOADING=1" instead of "RELOAD=1" to follow the format
expected by systemd (see sd_notify(3) man page).
Do not send STOPPING=1 in remote console mode:
attempting to execute "asterisk -rx" by the main process leads to
a warning if NotifyAccess=main (the default) or to a forced termination
if NotifyAccess=all.
(cherry picked from commit d3ff7c3eee)
Include signal.h to avoid the following build failure with uclibc-ng
raised since
2694792e13:
stasis/control.c: In function 'exec_command_on_condition':
stasis/control.c:313:3: warning: implicit declaration of function 'pthread_kill'; did you mean 'pthread_yield'? [-Wimplicit-function-declaration]
313 | pthread_kill(control->control_thread, SIGURG);
| ^~~~~~~~~~~~
| pthread_yield
stasis/control.c:313:41: error: 'SIGURG' undeclared (first use in this function)
313 | pthread_kill(control->control_thread, SIGURG);
| ^~~~~~
cherry-pick-to: 18
cherry-pick-to: 20
cherry-pick-to: 21
Fixes: #729
(cherry picked from commit c47307567a)
Currently, reloading res_pjsip will cause logging
to be disabled. This is because logging can also
be controlled via the debug option in pjsip.conf
and this defaults to "no".
To improve this, logging is no longer disabled on
reloads if logging had not been previously
enabled using the debug option from the config.
This ensures that logging enabled from the CLI
will persist through a reload.
ASTERISK-29912 #close
Resolves: #246
UserNote: Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
(cherry picked from commit 9fc596aaa7)
Add unique verbose prefixes for levels higher than 4, so
that these can be visually differentiated from each other.
Resolves: #721
(cherry picked from commit 57bb09667d)
Some of the money announcements can be off by one cent,
due to the use of floating point in the money calculations,
which is bad for obvious reasons.
This replaces floating point with simple string parsing
to ensure the cents value is converted accurately.
Resolves: #525
(cherry picked from commit 3a49f9ade5)
Because of the (often recursive) nature of module dependencies in
Asterisk, hot swapping a module on the fly is cumbersome if a module
is depended on by other modules. Currently, dependencies must be
popped manually by unloading dependents, unloading the module of
interest, and then loading modules again in reverse order.
To make this easier, the ability to do this recursively in certain
circumstances has been added, as an optional extension to the
"module refresh" command. If requested, Asterisk will check if a module
that has a positive usecount could be unloaded safely if anything
recursively dependent on it were unloaded. If so, it will go ahead
and unload all these modules and load them back again. This makes
hot swapping modules that provide dependencies much easier.
Resolves: #474
UserNote: In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
(cherry picked from commit a056e94885)
First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation.
Fixes#710
(cherry picked from commit 1423cfee29)
If the hostname field of the ast_tcptls_session_args structure is
set (which it is for websocket client connections), that hostname
will now automatically be used in an SNI TLS extension in the client
hello.
Resolves: #713
UserNote: Secure websocket client connections now send SNI in
the TLS client hello.
(cherry picked from commit 4d6f84a14f)
* Fixed possible memory leak in tn_config:tn_get_etn() where we
weren't releasing etn if tn or eprofile were null.
* We now canonicalize TNs before using them for lookups or adding
them to Identity headers.
* Fixed a typo in stir_shaken.conf.sample.
Resolves: #716
(cherry picked from commit b7ed77a7c5)
Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must
always be included in buildopts.h instead of only when
ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise.
Resolves: #719
(cherry picked from commit d1d80af0c0)
Fixed crash error when cli "module reload". The error appears when
compiling with res_prometheus and using the sorcery memory cache for
registrations
(cherry picked from commit b8525e87ae)
Commit f2f397c1a8 previously
made it possible to send Caller ID parameters to FXS stations
which, prior to that, could not be sent.
This change is complementary in that we now handle receiving
all these parameters on FXO lines and provide these up to
the dialplan, via chan_dahdi. In particular:
* If a redirecting reason is provided, the channel's redirecting
reason is set. No redirecting number is set, since there is
no parameter for this in the Caller ID protocol, but the reason
can be checked to determine if and why a call was forwarded.
* If the Call Qualifier parameter is received, the Call Qualifier
variable is set.
* Some comments have been added to explain why some of the code
is the way it is, to assist other people looking at it.
With this change, Asterisk's Caller ID implementation is now
reasonably complete for both FXS and FXO operation.
Resolves: #681
(cherry picked from commit 4cf8d9d94a)
If you're tracing a large function that may call another function
multiple times in different circumstances, it can be difficult to
see from the trace output exactly which location that function
was called from. There's no good way to automatically determine
the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT
simply print out a trace line before and after the call.
The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is
that SCOPE_CALL ignores the function's return value (if any) where
SCOPE_CALL_WITH_RESULT allows you to specify the type of the
function's return value so it can be assigned to a variable.
SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT
and the "int" return type.
(cherry picked from commit cf5c46d8ae)
The existing code sets the queue strategy to `ringall` but it is then
immediately overwritten with an invalid one.
Fixes#707
(cherry picked from commit 6914c93791)
rtp_engine.c and stun.c were calling ast_register_cleanup which
is skipped if any loadable module can't be cleanly unloaded
when asterisk shuts down. Since this will always be the case,
their cleanup functions never get run. In a practical sense
this makes no difference since asterisk is shutting down but if
you're in development mode and trying to use the leak sanitizer,
the leaks from both of those modules clutter up the output.
(cherry picked from commit 1eec3b6d18)
Emit a warning if REDIRECTING(reason) is set to an invalid
reason, consistent with what happens when
REDIRECTING(orig-reason) is set to an invalid reason.
Resolves: #683
(cherry picked from commit 9ba6875a3f)
* Add an AMI action to correspond to the "dahdi show status"
command, allowing span information to be retrieved via AMI.
* Show span number and sig type in "dahdi show channels".
Resolves: #673
(cherry picked from commit 4ff93474df)
Add a new identify_by option to res_pjsip_endpoint_identifier_ip
called 'transport' this matches endpoints based on the bound
ip address (local) instead of the 'ip' option, which matches on
the source ip address (remote).
UserNote: set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
(cherry picked from commit c8769f3d5a)
* OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use
X509_get_pubkey. The difference is that X509_get_pubkey requires
the caller to free the EVP_PKEY themselves so we now let
RAII_VAR do that.
* OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now
wrap it in an ao2 object.
* OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all
the certs from an X509_STORE and there's no easy way to polyfill
it so the CLI commands that list profiles will show a "not
supported" message instead of listing the certs in a store.
Resolves: #676
(cherry picked from commit 16b264d6a9)
There were a few references in the embedded documentation XML
where the case didn't match or where the referenced app or function
simply didn't exist any more. These were causing 404 responses
in docs.asterisk.org.
(cherry picked from commit 9e2179baa1)
Add ability to match against PJSIP request URI.
UserNote: this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
(cherry picked from commit ac297d15f8)
Commit 729cb1d390 added logic to retry
opening DAHDI channels on "dahdi restart" if they failed initially,
up to 1,000 times in a loop, to address cases where the channel was
still in use. However, this retry loop does not use the actual error,
which means chan_dahdi will also retry opening nonexistent channels
1,000 times per channel, causing a flood of unnecessary warning logs
for an operation that will never succeed, with tens or hundreds of
thousands of open attempts being made.
The original patch would have been more targeted if it only retried
on the specific relevant error (likely EBUSY, although it's hard to
say since the original issue is no longer available).
To avoid the problem above while avoiding the possibility of breakage,
this skips the retry logic if the error is ENXIO (No such device or
address), since this will never succeed.
Resolves: #669
(cherry picked from commit 63aa08fa0b)
This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary.
Fixes#657
(cherry picked from commit 555eb9d3d2)
There was functionality in chan_sip to get REFER headers, with GET_TRANSFERRER_DATA variable. This commit implements the same functionality in pjsip, to ease transfer from chan_sip to pjsip.
Fixes: #579
UserNote: the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
(cherry picked from commit cba82273ae)
SQLAlchemy 2.0 changed the way that commits/rollbacks are handled
causing the final `UPDATE` to our `alembic_version_<whatever>` tables
to be rolled back instead of committed.
We now use one connection to determine which
`alembic_version_<whatever>` table to use and another to run the
actual migrations. This prevents the erroneous rollback.
This change is compatible with both SQLAlchemy 1.4 and 2.0.
(cherry picked from commit 1944c9d72e)
manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
The action originate does not has the ability to run an subroutine at initial channel, like the Aplication Originate. This update give this ability for de action originate too.
For example, we can run a routine via Gosub on the channel to request an automatic answer, so the caller does not need to accept the call when using the originate command via manager, making the operation more efficient.
UserNote: When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
(cherry picked from commit d4b79cb466)
It is possible for dialplan to result in an infinite
recursion of variable substitution, which eventually
leads to stack overflow. If we detect this, abort
substitution and log an error for the user to fix
the broken dialplan.
Resolves: #480
UpgradeNote: The maximum amount of dialplan recursion
using variable substitution (such as by using EVAL_EXTEN)
is capped at 15.
(cherry picked from commit 310c8a7c68)
The prometheus exposition format requires each line to be unique[1].
This is handled by struct prometheus_metric having a list of children
that is managed when registering a metric. In case the scrape callback
is used, it is the responsibility of the implementation to handle this
correctly.
Originally the bridge callback didn't handle NULL snapshots, the crash
fix lead to NULL metrics, and fixing that lead to duplicates.
The original code assumed that snapshots are not NULL and then relied on
"if (i > 0)" to establish the parent/children relationship between
metrics of the same class. This is not workerable as the first bridge
might be invisible/lacks a snapshot.
Fix this by keeping a separate array of the first metric by class.
Instead of relying on the index of the bridge, check whether the array
has an entry. Use that array for the output.
Add a test case that verifies that the help text is not duplicated.
Resolves: #642
[1] https://prometheus.io/docs/instrumenting/exposition_formats/#grouping-and-sorting
(cherry picked from commit d45c8e165f)
This adds a CLI command that can be used to manually
kick specific AMI sessions.
Resolves: #485
UserNote: The "manager kick session" CLI command now
allows kicking a specified AMI session.
(cherry picked from commit f4fba80708)
The existing "waitfordialtone" setting in chan_dahdi.conf
applies permanently to a specific channel, regardless of
how it is being used. This rather restrictively prevents
a system from simultaneously being able to pick free lines
for outgoing calls while also allowing barge-in to a trunk
by some other arrangement.
This allows specifying "waitfordialtone" using the CHANNEL
function for only the next call that will be placed, allowing
significantly more flexibility in the use of trunk interfaces.
Resolves: #472
UserNote: "waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
(cherry picked from commit 7f51313725)
Currently, if a parking lot is full, bridge setup returns -1,
causing dialplan execution to terminate without TryExec.
However, such failures should be handled more gracefully,
the same way they are on other paths, as indicated by the
module's author, here:
http://lists.digium.com/pipermail/asterisk-dev/2018-December/077144.html
Now, callers will hear the parking failure announcement, and dialplan
will continue, which is consistent with existing failure modes.
Resolves: #624
(cherry picked from commit e1dfa20797)
In handle_negotiated_sdp the pending_media_state->read_callbacks must be
reset before they are added in the SDP handlers in
handle_negotiated_sdp_session_media. Otherwise, old callbacks for
removed streams and file descriptors could be added to the channel and
Asterisk would poll on non-existing file descriptors.
Resolves: #611
(cherry picked from commit c5a6d8a6db)
* Added checks for missing session, session->channel and rdata
in stir_shaken_incoming_request.
* Added checks for missing session, session->channel and tdata
in stir_shaken_outgoing_request.
Resolves: #645
Add a timeout option to control the amount of time
to wait if no early media is received before giving
up. This allows aborting early if the destination
is not being responsive.
Resolves: #588
UserNote: The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.
(cherry picked from commit bdf9327a33)
Most app_voicemail unit tests were not properly cleaning up
after themselves after running. This led to test mailboxes
lingering around in the system. It also meant that if any
unit tests in app_voicemail that create mailboxes were executed
and the module was not unloaded/loaded again prior to running
the test_voicemail_vm_info unit test, Asterisk would segfault
due to an attempt to copy a NULL string.
The load_config test did actually have logic to reinitialize
the config after the test. However, this did not work in practice
since load_config() would not reload the config since voicemail.conf
had not changed during the test; thus, additional logic has been
added to ensure that voicemail.conf is truly reloaded, after any
unit tests which modify the users list.
This prevents the SEGV due to invalid mailboxes lingering around,
and also ensures that the system state is restored to what it was
prior to the tests running.
Resolves: #629
(cherry picked from commit 2ac9c8fb5c)
