Naveen Albert [Thu, 21 Mar 2024 15:25:17 +0000 (11:25 -0400)]
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.
Sean Bright [Thu, 7 Mar 2024 17:27:12 +0000 (12:27 -0500)]
res_monitor.c: Don't emit a warning about 'X' being unrecognized.
Code was added in 030f7d41 to warn if an unrecognized option was
passed to an application, but code in Monitor was taking advantage of
the fact that the application would silently accept an invalid option.
We now recognize the invalid option but we don't do anything if it's
set.
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 17:45:34 +0000 (12:45 -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.
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.
Naveen Albert [Sat, 23 Dec 2023 16:26:13 +0000 (16:26 +0000)]
app_if: Fix faulty EndIf branching.
This fixes faulty branching logic for the
EndIf application. Instead of computing
the next priority, which should be done
for false conditionals or ExitIf, we should
simply advance to the next priority.
Naveen Albert [Tue, 26 Dec 2023 17:14:11 +0000 (17:14 +0000)]
manager.c: Fix regression due to using wrong free function.
Commit 424be345639d75c6cb7d0bd2da5f0f407dbd0bd5 introduced
a regression by calling ast_free on memory allocated by
realpath. This causes Asterisk to abort when executing this
function. Since the memory is allocated by glibc, it should
be freed using ast_std_free.
Naveen Albert [Tue, 5 Dec 2023 16:36:47 +0000 (11:36 -0500)]
manager.c: Improve clarity of "manager show connected".
Improve the "manager show connected" CLI command
to clarify that the last two columns are permissions
related, not counts, and use sufficient widths
to consistently display these values.
Although `make_xml_documentation`'s `print_dependencies` command was
corrected by the previous fix (#461) for #142, the `create_xml` was
not properly handling `LOCAL_MOD_SUBDIRS` XML documentation.
Naveen Albert [Thu, 9 Nov 2023 21:26:46 +0000 (16:26 -0500)]
general: Fix broken links.
This fixes a number of broken links throughout the
tree, mostly caused by wiki.asterisk.org being replaced
with docs.asterisk.org, which should eliminate the
need for sporadic fixes as in f28047db36a70e81fe373a3d19132c43adf3f74b.
app_dial: Add option "j" to preserve initial stream topology of caller
Resolves: #462
UserNote: The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
George Joseph [Sun, 12 Nov 2023 00:40:10 +0000 (17:40 -0700)]
ast_coredumper: Increase reliability
Instead of searching for the asterisk binary and the modules in the
filesystem, we now get their locations, along with libdir, from
the coredump itself...
For the binary, we can use `gdb -c <coredump> ... "info proc exe"`.
gdb can print this even without having the executable and symbols.
Once we have the binary, we can get the location of the modules with
`gdb ... "print ast_config_AST_MODULE_DIR`
If there was no result then either it's not an asterisk coredump
or there were no symbols loaded. Either way, it's not usable.
For libdir, we now run "strings" on the note0 section of the
coredump (which has the shared library -> memory address xref) and
search for "libasteriskssl|libasteriskpj", then take the dirname.
Since we're now getting everything from the coredump, it has to be
correct as long as we're not crossing namespace boundaries like
running asterisk in a docker container but trying to run
ast_coredumper from the host using a shared file system (which you
shouldn't be doing).
There is still a case for using --asterisk-bin and/or --libdir: If
you've updated asterisk since the coredump was taken, the binary,
libraries and modules won't match the coredump which will render it
useless. If you can restore or rebuild the original files that
match the coredump and place them in a temporary directory, you can
use --asterisk-bin, --libdir, and a new --moddir option to point to
them and they'll be correctly captured in a tarball created
with --tarball-coredumps. If you also use --tarball-config, you can
use a new --etcdir option to point to what normally would be the
/etc/asterisk directory.
Sean Bright [Fri, 1 Dec 2023 17:03:29 +0000 (12:03 -0500)]
logger.c: Move LOG_GROUP documentation to dedicated XML file.
The `get_documentation` awk script will only extract the first
DOCUMENTATION block that it finds in a given file. This is by design
(9bc2127) to prevent AMI event documentation from being pulled in to
the core.xml documentation file.
Because of this, the `LOG_GROUP` documentation added in 89709e2 was
not being properly extracted and was missing fom the resulting XML
documentation file. This commit moves the `LOG_GROUP` documentation to
a separate `logger.xml` file.
res_odbc.c: Allow concurrent access to request odbc connections
There are valid scenarios where res_odbc's connection pool might have some dead
or stuck connections while others are healthy (imagine network
elements/firewalls/routers silently timing out connections to a single DB and a
single IP address, or a heterogeneous connection pool connected to potentially
multiple IPs/instances of a replicated DB using a DNS front end for load
balancing and one replica fails).
In order to time out those unhealthy connections without blocking access to
other parts of Asterisk that may attempt access to the connection pool, it would
be beneficial to not lock/block access around the entire pool in
_ast_odbc_request_obj2 while doing potentially blocking operations on connection
pool objects such as the connection_dead() test, odbc_obj_connect(), or by
dereferencing a struct odbc_obj for the last time and triggering a
odbc_obj_disconnect().
This would facilitate much quicker and concurrent timeout of dead connections
via the connection_dead() test, which could block potentially for a long period
of time depending on odbc.ini or other odbc connector specific timeout settings.
This also would make rapid failover (in the clustered DB scenario) much quicker.
This patch changes the locking in _ast_odbc_request_obj2() to not lock around
odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to
lock around truly shared, non-immutable state like the connection_cnt member and
the connections list on struct odbc_class.
Sean Bright [Wed, 22 Nov 2023 19:25:19 +0000 (14:25 -0500)]
config.c: Log #exec include failures.
If the script referenced by `#exec` does not exist, writes anything to
stderr, or exits abnormally or with a non-zero exit status, we log
that to Asterisk's error logging channel.
Additionally, write out a warning if the script produces no output.
Resequencing is a process that occurs when we open a voicemail folder
and discover that there are gaps between messages (e.g. `msg0000.txt`
is missing but `msg0001.txt` exists). Resequencing involves shifting
the existing messages down so we end up with a sequential list of
messages.
Currently, this process stops after reaching a threshold based on the
message limit (`maxmsg`) configured on the current folder. However, if
`maxmsg` is lowered when a voicemail folder contains more than
`maxmsg + 10` messages, resequencing will not run completely leaving
the mailbox in an inconsistent state.
We now resequence up to the maximum number of messages permitted by
`app_voicemail` (currently hard-coded at 9999 messages).
Naveen Albert [Fri, 24 Nov 2023 13:06:58 +0000 (08:06 -0500)]
sig_analog: Fix channel leak when mwimonitor is enabled.
When mwimonitor=yes is enabled for an FXO port,
the do_monitor thread will launch mwi_thread if it thinks
there could be MWI on an FXO channel, due to the noise
threshold being satisfied. This, in turns, calls
analog_ss_thread_start in sig_analog. However, unlike
all other instances where __analog_ss_thread is called
in sig_analog, this call path does not properly set
pvt->ss_astchan to the Asterisk channel, which means
that the Asterisk channel is NULL when __analog_ss_thread
starts executing. As a result, the thread exits and the
channel is never properly cleaned up by calling ast_hangup.
This caused issues with do_monitor on incoming calls,
as it would think the channel was still owned even while
receiving events, leading to an infinite barrage of
warning messages; additionally, the channel would persist
improperly.
To fix this, the assignment is added to the call path
where it is missing (which is only used for mwi_thread).
A warning message is also added since previously there
was no indication that __analog_ss_thread was exiting
abnormally. This resolves both the channel leak and the
condition that led to the warning messages.
Certain channel options are not set anywhere or
exposed in any way to users, making them unusable.
This exposes some of these options which make sense
for users to manipulate at runtime.
Sean Bright [Tue, 7 Nov 2023 20:03:53 +0000 (15:03 -0500)]
app.c: Allow ampersands in playback lists to be escaped.
Any function or application that accepts a `&`-separated list of
filenames can now include a literal `&` in a filename by wrapping the
entire filename in single quotes, e.g.:
UpgradeNote: Ampersands in URLs passed to the `Playback()`,
`Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
`Queue()` applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the `CONFBRIDGE` dialplan function, or configuring various
features in `confbridge.conf` and `queues.conf`.
Sean Bright [Thu, 9 Nov 2023 14:11:33 +0000 (09:11 -0500)]
live_ast: Add astcachedir to generated asterisk.conf.
`astcachedir` (added in b0842713) was not added to `live_ast` so
continued to point to the system `/var/cache` directory instead of the
one in the live environment.
app_followme.c: Grab reference on nativeformats before using it
Fixes a crash due to a lack of proper reference on the nativeformats
object before passing it into ast_request(). Also found potentially
similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c
Naveen Albert [Thu, 9 Nov 2023 15:12:15 +0000 (10:12 -0500)]
configs: Improve documentation for bandwidth in iax.conf.
This improves the documentation for the bandwidth setting
in iax.conf by making it clearer what the ramifications
of this setting are. It also changes the sample default
from low to high, since only high is compatible with good
codecs that people will want to use in the vast majority
of cases, and this is a common gotcha that trips up new users.
Naveen Albert [Wed, 9 Aug 2023 22:12:55 +0000 (22:12 +0000)]
logger: Add channel-based filtering.
This adds the ability to filter console
logging by channel or groups of channels.
This can be useful on busy systems where
an administrator would like to analyze certain
calls in detail. A dialplan function is also
included for the purpose of assigning a channel
to a group (e.g. by tenant, or some other metric).
ASTERISK-30483 #close
Resolves: #242
UserNote: The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
George Joseph [Tue, 7 Nov 2023 23:24:37 +0000 (16:24 -0700)]
codec_ilbc: Disable system ilbc if version >= 3.0.0
Fedora 37 started shipping ilbc 3.0.4 which we don't yet support.
configure.ac now checks the system for "libilbc < 3" instead of
just "libilbc". If true, the system version of ilbc will be used.
If not, the version included at codecs/ilbc will be used.
George Joseph [Tue, 31 Oct 2023 21:08:14 +0000 (15:08 -0600)]
chan_pjsip: Add PJSIPHangup dialplan app and manager action
See UserNote below.
Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.
Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
603. This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).
Also extracted the XML documentation to its own file since it was
almost as large as the code itself.
UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
Sean Bright [Mon, 6 Nov 2023 16:45:33 +0000 (11:45 -0500)]
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE
in a frame was one that may not have any data - such as the CALLTOKEN
IE in an NEW request - it was not getting displayed.
projects / thirdparty / asterisk.git / log
