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
Fixes: #388
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.
Resolves: #425
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.
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.
Resolves: #84
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.
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.
This adds optional ADSI support to the Directory
application, which allows callers with ADSI CPE
to navigate the Directory system significantly
faster than is possible using the audio prompts.
Callers can see the directory name (and optionally
extension) on their screenphone and confirm or
reject a match immediately rather than waiting
for it to be spelled out, enhancing usability.
Resolves: #356
Currently, trying to call a Local channel with a slash
in the extension will fail due to the parsing of characters
after such a slash as being dial modifiers. Additionally,
core_local is inconsistent and incomplete with
its parsing of Local dial strings in that sometimes it
uses the first slash and at other times it uses the last.
For instance, something like DAHDI/5 or PJSIP/device
is a perfectly usable extension in the dialplan, but Local
channels in particular prevent these from being called.
This creates inconsistent behavior for users, since using
a slash in an extension is perfectly acceptable, and using
a Goto to accomplish this works fine, but if specified
through a Local channel, the parsing prevents this.
This fixes this by explicitly parsing options from the
last slash in the extension, rather than the first one,
which doesn't cause an issue for extensions with slashes.
ASTERISK-30013 #close
Resolves: #248
This adds an AMI event that is emitted whenever a
mailbox password is successfully changed, allowing
AMI consumers to process these.
UserNote: The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
In simple_bridge_join, we were sending topology change requests
even when the new and old topologies were the same. In some
circumstances, this can cause unnecessary re-invites and even
a re-invite flood. We now suppress those.
Resolves: #384
If too many ciphers are specified in the PJSIP config,
include the maximum number of ciphers that may be
specified in the user-facing error message.
Resolves: #396
* Allow res_speech to translate the input channel if the
format is translatable to a format suppored by the
speech provider.
Resolves: #129
UserNote: res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
The '*' list indicator for default values and allowable values for
path, query and POST parameters need to be indented 4 spaces
instead of 2.
Should resolve issue 38 in the documentation repo.
Per RFC8827:
Implementations MUST NOT implement DTLS renegotiation and MUST
reject it with a "no_renegotiation" alert if offered.
So we disable it when webrtc=yes is set.
Fixes#378
UpgradeNote: The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
Commit f66f77f last year prevents the res_pjsip_exten_state and
res_pjsip_mwi modules from unloading due to possible pjproject
asserts if the modules are reloaded. A side effect of the
implementation is that the taskprocessors these modules use aren't
being released. When asterisk is doing a graceful shutdown, it
waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all
taskprocessors to stop but since those 2 modules don't release
theirs, the shutdown hangs for that amount of time.
This change allows the modules to be unloaded and their resources to
be released when ast_shutdown_final is true.
Resolves: #379
This commit introduces an extension to the endpoint and relevant
resource sizes for PJSIP, transitioning from its current 40-character
constraint to a more versatile 255-character capacity. This enhancement
significantly overcomes limitations related to domain qualification and
practical usage, ultimately delivering improved functionality. In
addition, it includes adjustments to accommodate the expanded realm size
within the ARI, specifically enhancing the maximum realm length.
Resolves: #345
UserNote: With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
UpgradeNote: As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
res_statsis's app loop sleeps for up to .2s waiting on input
to a channel before re-checking the command queue. This can
cause delays between channel setup and bridge.
This change is to send a SIGURG on the sleeping thread when
a new command is enqueued. This exits the sleeping thread out
of the ast_waitfor() call triggering the new command being
processed on the channel immediately.
Resolves: #362
UserNote: Call setup times should be significantly improved
when using ARI.
Make it possible to start a playback and the calling party
to receive audio on a bridge before the call is connected.
Model the implementation after play_on_channel and deliver a
AST_CONTROL_PROGRESS before starting the playback.
For a PJSIP channel this will result in sending a SIP 183
Session Progress.
You can now define the _TRACE_PREFIX_ macro to change the
default trace line prefix of "file:line function" to
something else. Full documentation in logger.h.
The current STIR/SHAKEN implementation is not currently usable due
to encryption issues. Rather than trying to futz with OpenSSL and
the the current code, we can take advantage of the existing
capabilities of libjwt but we first need to add it to the
third-party infrastructure already in place for jansson and
pjproject.
A few tweaks were also made to the third-party infrastructure as
a whole. The jansson "dest" install directory was renamed "dist"
to better match convention, and the third-party Makefile was updated
to clean all product directories not just the ones currently in
use.
Resolves: #349
The documentation on qualify_timeout does not explicitly state that the timeout
includes any time required to perform any needed DNS queries on the endpoint.
If the OPTIONS response is delayed due to the DNS query, it can still render an
endpoint as Unreachable if the net time is enough for qualify_timeout to expire.
Resolves: #352
Internally, chan_dahdi only applies callgroup and
pickupgroup to FXO signalled channels, but this is
not documented anywhere. This is now documented in
the sample config, and a warning is emitted if a
user tries configuring these settings for channel
types that do not support these settings, since they
will not have any effect.
Resolves: #294
This commit fixes crashes in JSON_DECODE() for types null, true, false
and real numbers.
In addition it ensures that a path is not deeper than 32 levels.
Also allow root object to be an array.
Add unit tests for above cases.
res_speech_aeap previously did not register an error handler
with aeap, so it was not notified of a disconnect. This resulted
in SpeechBackground never exiting upon a websocket disconnect.
Resolves: #303
If ADSI is available on a channel, app_voicemail will repeatedly
try to use ADSI, even if there is no CPE that supports it. This
leads to many unnecessary delays during the session. If ADSI is
available but ADSI setup fails, we now disable it to prevent
further attempts to use ADSI during the session.
Resolves: #354
Some providers require a multiple of 20 for the maxptime or fail to complete calls,
e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used.
Resolves: #260
Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
causes the lock calls to loop over trylock in 200us intervals until
the lock is obtained and spits out log messages if it takes more
than 5 seconds. From a code perspective, the only reason they were
tied together was for logging. So... The ifdefs in lock.c were
refactored to allow DETECT_DEADLOCKS to be enabled without
also enabling DEBUG_THREADS.
Resolves: #321
UserNote: You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
The CLI .asterisk_history file is read from/written to the directory
specified by the HOME environment variable. If the root user starts
asterisk with the -U/-G options, or with runuser/rungroup set in
asterisk.conf, the asterisk process is started as root but then it
calls setuid/setgid to set the new user/group. This does NOT reset
the HOME environment variable to the new user's home directory
though so it's still left as "/root". In this case, the new user
will almost certainly NOT have access to read from or write to the
history file.
* Added function process_histfile() which calls
getpwuid(geteuid()) and uses pw->dir as the home directory
instead of the HOME environment variable.
* ast_el_read_default_histfile() and ast_el_write_default_histfile()
have been modified to use the new process_histfile()
function.
Resolves: #337
From the gdb information, ast_websocket_read reads a message successfully,
then transport_read is called in the serializer. During execution of pjsip_transport_down,
ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop.
After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages.
This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue.
In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop.
Resolves: asterisk#299
