In 5ac5c2b0 we defined `OPENSSL_SUPPRESS_DEPRECATED` to silence
deprecation warnings. This commit switches over to using
non-deprecated API.
(cherry picked from commit 3859b630a7)
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.
Resolves: #442
(cherry picked from commit c222343ec6)
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.:
```
exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world)
```
Fixes#172
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`.
(cherry picked from commit f8212d4594)
I assume this was missed when initially converting to XML
documentation and we've been kicking the can down the road since.
(cherry picked from commit 9e1a60727e)
Additionally add a `assert()` to in the TLS client setup code to
ensure that hostname is set when it is supposed to be.
Fixes#433
(cherry picked from commit f2961f048d)
`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.
(cherry picked from commit 978d09fc35)
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
(cherry picked from commit 275f7911b5)
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
(cherry picked from commit 1f19227eab)
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.
(cherry picked from commit a0fc8d1b5f)
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
(cherry picked from commit d819a6bccb)
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.
(cherry picked from commit cd77953172)
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.
(cherry picked from commit 0e126b3841)
If attempting to ring a channel using a nonexistent cadence,
emit a warning, before falling back to the default cadence.
Resolves: #409
(cherry picked from commit 4b9a4483fc)
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
(cherry picked from commit 5046620fa3)
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
(cherry picked from commit 2191a0d33f)
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
(cherry picked from commit cf9d7fa9f6)
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
(cherry picked from commit f301d4559e)
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
(cherry picked from commit 8d9d0d3738)
* 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.
(cherry picked from commit c3e2bff36e)
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.
(cherry picked from commit b619b64137)
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.
(cherry picked from commit 9d329da346)
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
(cherry picked from commit 9efc4bdfbc)
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.
(cherry picked from commit f89e56c178)
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.
(cherry picked from commit 289aa1840e)
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.
(cherry picked from commit 0c1c6e9ada)
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.
(cherry picked from commit 79220e3f0c)
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.
(cherry picked from commit 1171dcee02)
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.
(cherry picked from commit b52e07ee1b)
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
(cherry picked from commit a9d4175e1d)