Mike Bradeen [Mon, 8 Apr 2024 17:00:14 +0000 (11:00 -0600)]
rtp_engine: add support for multirate RFC2833 digits
Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.
Asterisk currently treats RFC2833 Digits as a single rtp payload type
with a fixed bitrate of 8K. This change would expand that to 8, 16,
24 and 32K.
This requires checking the offered rtp types for any of these bitrates
and then adding an offer for each (if configured for RFC2833.) DTMF
generation must also be changed in order to look at the current outbound
codec in order to generate appropriately timed rtp.
For cases where no outgoing audio has yet been sent prior to digit
generation, Asterisk now has a concept of a 'preferred' codec based on
offer order.
On inbound calls Asterisk will mimic the payload types of the RFC2833
digits.
On outbound calls Asterisk will choose the next free payload types starting
with 101.
UserNote: No change in configuration is required in order to enable this
feature. Endpoints configured to use RFC2833 will automatically have this
enabled. If the endpoint does not support this, it should not include it in
the SDP offer/response.
Ivan Poddubny [Sun, 5 May 2024 12:53:11 +0000 (14:53 +0200)]
asterisk.c: Fix sending incorrect messages to systemd notify
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.
Include signal.h to avoid the following build failure with uclibc-ng
raised since
https://github.com/asterisk/asterisk/commit/2694792e13c7f3ab1911c4a69fba0df32c544177:
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);
| ^~~~~~
Naveen Albert [Wed, 9 Aug 2023 22:41:24 +0000 (22:41 +0000)]
res_pjsip_logger: Preserve logging state on reloads.
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.
Naveen Albert [Sat, 2 Dec 2023 23:07:02 +0000 (18:07 -0500)]
loader.c: Allow dependent modules to be unloaded recursively.
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.
Henrik Liljedahl [Thu, 11 Apr 2024 10:56:55 +0000 (12:56 +0200)]
res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeout setting.
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.
George Joseph [Tue, 23 Apr 2024 20:15:20 +0000 (14:15 -0600)]
tcptls/iostream: Add support for setting SNI on client TLS connections
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.
George Joseph [Thu, 25 Apr 2024 17:56:15 +0000 (11:56 -0600)]
stir_shaken: Fix memory leak, typo in config, tn canonicalization
* 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.
George Joseph [Sat, 27 Apr 2024 20:40:28 +0000 (14:40 -0600)]
make_buildopts_h: Always include DETECT_DEADLOCKS
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.
Naveen Albert [Mon, 1 Apr 2024 21:16:29 +0000 (17:16 -0400)]
callerid.c: Parse previously ignored Caller ID parameters.
Commit f2f397c1a8cc48913434ebb297f0ff50d96993db 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.
George Joseph [Tue, 9 Apr 2024 20:49:36 +0000 (14:49 -0600)]
logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
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.
George Joseph [Tue, 2 Apr 2024 20:28:35 +0000 (14:28 -0600)]
rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
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.
Naveen Albert [Mon, 1 Apr 2024 21:24:13 +0000 (17:24 -0400)]
func_callerid: Emit warning if invalid redirecting reason set.
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.
Naveen Albert [Fri, 29 Mar 2024 14:46:18 +0000 (10:46 -0400)]
chan_dahdi: Add DAHDIShowStatus AMI action.
* 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".
Sperl Viktor [Thu, 28 Mar 2024 15:22:24 +0000 (16:22 +0100)]
res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
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.
George Joseph [Mon, 1 Apr 2024 20:10:32 +0000 (14:10 -0600)]
res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2)
* 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.
George Joseph [Mon, 1 Apr 2024 12:30:54 +0000 (06:30 -0600)]
Fix incorrect application and function documentation references
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.
Sperl Viktor [Thu, 28 Mar 2024 13:20:26 +0000 (14:20 +0100)]
res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
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.
Naveen Albert [Tue, 26 Mar 2024 11:43:32 +0000 (07:43 -0400)]
chan_dahdi: Don't retry opening nonexistent channels on restart.
Commit 729cb1d390b136ccc696430aa5c68d60ea4028be 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.
Joshua Elson [Mon, 18 Mar 2024 19:14:36 +0000 (15:14 -0400)]
Implement Configurable TCP Keepalive Settings in PJSIP Transports
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.
Martin Tomec [Tue, 6 Feb 2024 16:39:18 +0000 (17:39 +0100)]
res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
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 cba82273ae789c193128589a6c3f3679782b1ccd)
Sean Bright [Wed, 20 Mar 2024 16:20:40 +0000 (12:20 -0400)]
alembic: Fix compatibility with SQLAlchemy 2.0+.
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.
jonatascalebe [Thu, 14 Mar 2024 18:53:43 +0000 (15:53 -0300)]
manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
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()
Naveen Albert [Mon, 4 Dec 2023 17:58:17 +0000 (12:58 -0500)]
pbx_variables.c: Prevent SEGV due to stack overflow.
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.
res_prometheus: Fix duplicate output of metric and help text
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.
Naveen Albert [Sat, 2 Dec 2023 18:24:20 +0000 (13:24 -0500)]
chan_dahdi: Allow specifying waitfordialtone per call.
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.
Naveen Albert [Sun, 3 Mar 2024 18:37:10 +0000 (13:37 -0500)]
res_parking: Fail gracefully if parking lot is full.
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:
Fixes: asterisk#648
UserNote: Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
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.
Naveen Albert [Thu, 8 Feb 2024 18:09:49 +0000 (13:09 -0500)]
app_dial: Add dial time for progress/ringing.
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.
Naveen Albert [Thu, 29 Feb 2024 14:27:09 +0000 (09:27 -0500)]
app_voicemail: Properly reinitialize config after unit tests.
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.
George Joseph [Mon, 4 Mar 2024 16:46:48 +0000 (09:46 -0700)]
Makefile: Add stir_shaken/cache to directories created on install
The default location for the stir_shaken cache is
/var/lib/asterisk/keys/stir_shaken/cache but we were only creating
/var/lib/asterisk/keys/stir_shaken on istall. We now create
the cache sub-directory.
George Joseph [Thu, 26 Oct 2023 16:27:35 +0000 (10:27 -0600)]
Stir/Shaken Refactor
Why do we need a refactor?
The original stir/shaken implementation was started over 3 years ago
when little was understood about practical implementation. The
result was an implementation that wouldn't actually interoperate
with any other stir-shaken implementations.
There were also a number of stir-shaken features and RFC
requirements that were never implemented such as TNAuthList
certificate validation, sending Reason headers in SIP responses
when verification failed but we wished to continue the call, and
the ability to send Media Key(mky) grants in the Identity header
when the call involved DTLS.
Finally, there were some performance concerns around outgoing
calls and selection of the correct certificate and private key.
The configuration was keyed by an arbitrary name which meant that
for every outgoing call, we had to scan the entire list of
configured TNs to find the correct cert to use. With only a few
TNs configured, this wasn't an issue but if you have a thousand,
it could be.
What's changed?
* Configuration objects have been refactored to be clearer about
their uses and to fix issues.
* The "general" object was renamed to "verification" since it
contains parameters specific to the incoming verification
process. It also never handled ca_path and crl_path
correctly.
* A new "attestation" object was added that controls the
outgoing attestation process. It sets default certificates,
keys, etc.
* The "certificate" object was renamed to "tn" and had it's key
change to telephone number since outgoing call attestation
needs to look up certificates by telephone number.
* The "profile" object had more parameters added to it that can
override default parameters specified in the "attestation"
and "verification" objects.
* The "store" object was removed altogther as it was never
implemented.
* We now use libjwt to create outgoing Identity headers and to
parse and validate signatures on incoming Identiy headers. Our
previous custom implementation was much of the source of the
interoperability issues.
* General code cleanup and refactor.
* Moved things to better places.
* Separated some of the complex functions to smaller ones.
* Using context objects rather than passing tons of parameters
in function calls.
* Removed some complexity and unneeded encapsuation from the
config objects.
Resolves: #351
Resolves: #46
UserNote: Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
UpgradeNote: The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
Sebastian Jennen [Sun, 25 Feb 2024 20:53:57 +0000 (21:53 +0100)]
translate.c: implement new direct comp table mode
The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio.
This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing).
- add new table mode
- hide the 999999 comp values, as these only indicate an issue with transcoding
- hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding)
romryz [Tue, 6 Feb 2024 13:57:32 +0000 (15:57 +0200)]
res_rtp_asterisk.c: Correct coefficient in MOS calculation.
Media Experience Score relies on incorrect pseudo_mos variable
calculation. According to forming an opinion section of the
documentation, calculation relies on ITU-T G.107 standard:
ITU-T G.107 Annex B suggests to calculate MOS with a coefficient
"seven times ten to the power of negative six", 7 * 10^(-6). which
would mean 6 digits after the decimal point. Current implementation
has 7 digits after the decimal point, which downrates the calls.
Naveen Albert [Fri, 9 Feb 2024 22:07:13 +0000 (17:07 -0500)]
dsp.c: Fix and improve potentially inaccurate log message.
If ast_dsp_process is called with a codec besides slin, ulaw,
or alaw, a warning is logged that in-band DTMF is not supported,
but this message is not always appropriate or correct, because
ast_dsp_process is much more generic than just DTMF detection.
This logs a more generic message in those cases, and also improves
codec-mismatch logging throughout dsp.c by ensuring incompatible
codecs are printed out.
George Joseph [Fri, 9 Feb 2024 16:15:13 +0000 (09:15 -0700)]
pjsip show channelstats: Prevent possible segfault when faxing
Under rare circumstances, it's possible for the original audio
session in the active_media_state default_session to be corrupted
instead of removed when switching to the t38/image media session
during fax negotiation. This can cause a segfault when a "pjsip
show channelstats" attempts to print that audio media session's
rtp statistics. In these cases, the active_media_state
topology is correctly showing only a single t38/image stream
so we now check that there's an audio stream in the topology
before attempting to use the audio media session to get the rtp
statistics.
George Joseph [Wed, 31 Jan 2024 17:46:28 +0000 (10:46 -0700)]
Reduce startup/shutdown verbose logging
When started with a verbose level of 3, asterisk can emit over 1500
verbose message that serve no real purpose other than to fill up
logs. When asterisk shuts down, it emits another 1100 that are of
even less use. Since the testsuite runs asterisk with a verbose
level of 3, and asterisk starts and stops for every one of the 700+
tests, the number of log messages is staggering. Besides taking up
resources, it also makes it hard to debug failing tests.
This commit changes the log level for those verbose messages to 5
instead of 3 which reduces the number of log messages to only a
handful. Of course, NOTICE, WARNING and ERROR message are
unaffected.
There's also one other minor change...
ast_context_remove_extension_callerid2() logs a DEBUG message
instead of an ERROR if the extension you're deleting doesn't exist.
The pjsip_config_wizard calls that function to clean up the config
and has been triggering that annoying error message for years.
Naveen Albert [Mon, 12 Feb 2024 17:39:18 +0000 (12:39 -0500)]
configure: Rerun bootstrap on modern platform.
The last time configure was run, it was run on a system that
did not enable -std=gnu11 by default, which meant that the
restrict qualifier would not be recognized on certain platforms.
This regenerates the configure files from running bootstrap.sh,
so that these should be recognized on all supported platforms.
Ben Ford [Mon, 5 Feb 2024 20:15:12 +0000 (14:15 -0600)]
Upgrade bundled pjproject to 2.14.
Fixes: #406
UserNote: Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
cmaj [Fri, 2 Feb 2024 17:27:31 +0000 (10:27 -0700)]
app_speech_utils.c: Allow partial speech results.
Adds 'p' option to SpeechBackground() application.
With this option, when the app timeout is reached,
whatever the backend speech engine collected will
be returned as if it were the final, full result.
(This works for engines that make partial results.)
Resolves: #572
UserNote: The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
Joshua C. Colp [Wed, 31 Jan 2024 14:03:28 +0000 (10:03 -0400)]
utils: Make behavior of ast_strsep* match strsep.
Given the scenario of passing an empty string to the
ast_strsep functions the functions would return NULL
instead of an empty string. This is counter to how
strsep itself works.
This change alters the behavior of the functions to
match that of strsep.
Mike Bradeen [Wed, 31 Jan 2024 15:55:04 +0000 (08:55 -0700)]
app_chanspy: Add 'D' option for dual-channel audio
Adds the 'D' option to app chanspy that causes the input and output
frames of the spied channel to be interleaved in the spy output frame.
This allows the input and output of the spied channel to be decoded
separately by the receiver.
If the 'o' option is also set, the 'D' option is ignored as the
audio being spied is inherently one direction.
Fixes: #569
UserNote: The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
Naveen Albert [Sun, 28 Jan 2024 13:57:47 +0000 (08:57 -0500)]
app_if: Fix next priority calculation.
Commit fa3922a4d28860d415614347d9f06c233d2beb07 fixed
a branching issue but "overshoots" when calculating
the next priority. This fixes that; accompanying
test suite tests have also been extended.
Sean Bright [Sat, 27 Jan 2024 14:46:27 +0000 (09:46 -0500)]
res_pjsip_session.c: Correctly format SDP connection addresses.
Resolves a regression identified by @justinludwig involving the
rendering of IPv6 addresses in outgoing SDP.
Also updates `media_address` on PJSIP endpoints so that if we are able
to parse the configured value as an IP we store it in a format that we
can directly use later. Based on my reading of the code it appeared
that one could configure `media_address` as:
```
[foo]
type = endpoint
...
media_address = [2001:db8::]
```
And that value would be blindly copied into the outgoing SDP without
regard to its format.
Naveen Albert [Thu, 25 Jan 2024 17:44:29 +0000 (12:44 -0500)]
manager.c: Fix erroneous reloads in UpdateConfig.
Currently, a reload will always occur if the
Reload header is provided for the UpdateConfig
action. However, we should not be doing a reload
if the header value has a falsy value, per the
documentation, so this makes the reload behavior
consistent with the existing documentation.
Sean Bright [Sun, 21 Jan 2024 16:40:19 +0000 (11:40 -0500)]
app_confbridge: Don't emit warnings on valid configurations.
The numeric bridge profile options `internal_sample_rate` and
`maximum_sample_rate` are documented to accept the special values
`auto` and `none`, respectively. While these values currently work,
they also emit warnings when used which could be confusing for users.
In passing, also ensure that we only accept the documented range of
sample rate values between 8000 and 192000.
Mike Bradeen [Wed, 10 Jan 2024 16:47:24 +0000 (09:47 -0700)]
app_voicemail_odbc: remove macrocontext from voicemail_messages table
When app_macro was deprecated, the macrocontext column was removed from
the INSERT statement but the binds were not renumbered. This broke the
insert.
This change removes the macrocontext column via alembic and re-numbers
the existing columns in the INSERT.
Fixes: #527
UserNote: The fix requires removing the macrocontext column from the
voicemail_messages table in the voicemail database via alembic upgrade.
UpgradeNote: The fix requires that the voicemail database be upgraded via
alembic. Upgrading to the latest voicemail database via alembic will
remove the macrocontext column from the voicemail_messages table.
Naveen Albert [Fri, 10 Nov 2023 14:44:25 +0000 (09:44 -0500)]
chan_dahdi: Allow MWI to be manually toggled on channels.
This adds a CLI command to manually toggle the MWI status
of a channel, useful for troubleshooting or resetting
MWI devices, similar to the capabilities offered with
SIP messaging to manually control MWI status.
UserNote: The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Naveen Albert [Tue, 16 Jan 2024 21:58:22 +0000 (16:58 -0500)]
logger: Fix linking regression.
Commit 008731b0a4b96c4e6c340fff738cc12364985b64
caused a regression by resulting in logger.xml
being compiled and linked into the asterisk
binary in lieu of logger.c on certain platforms
if Asterisk was compiled in dev mode.
To fix this, we ensure the file has a unique
name without the extension. Most existing .xml
files have been named differently from any
.c files in the same directory or did not
pose this issue.
channels/pjsip/dialplan_functions.xml does not
pose this issue but is also being renamed
to adhere to this policy.
Over the past year, we've had several reports of "topology storms"
occurring where 2 external facing channels connected by one or more
local channels and bridges will get themselves in a state where
they continually send each other topology change requests. This
usually manifests itself in no-audio calls and a flood of
"Exceptionally long queue length" messages. It appears that this
commit is the cause so we're reverting it for now until we can
determine a more appropriate solution.
projects / thirdparty / asterisk.git / log
