Even though Dial() internally uses milliseconds for its dial timeouts,
this capability has been mostly obscured from users as the argument is
only parsed as an integer, thus forcing the use of whole seconds for
timeouts.
Parse it as a decimal instead so that timeouts can now truly have
millisecond precision.
Resolves: #1487
UserNote: The answer and progress dial timeouts now have millisecond
precision, instead of having to be whole numbers.
Commit dc8e3eeaaf improved the debug log
messages in dsp.c. This makes two minor corrections to it:
* Properly guard an added log statement in a conditional.
* Don't add one to the hit count if there was no hit (however, we do
still want to do this for the case where this is one).
Resolves: #1496
When running "config show help <module>", if no XML documentation exists
for the specified module, "Module <module> not found." is returned,
which is misleading if the module is loaded but simply has no XML
documentation for its config. Improve the message to clarify that the
module may simply have no config documentation.
Resolves: #1489
The previous lack of an example made it ambiguous if the arguments went
inside the function arguments or were part of the right-hand value.
Resolves: #1485
If Last Number Redial is used to redial, ensure that we do not wait
for further digits. This was possible if the number that was last
dialed is a prefix of another possible dialplan match. Since all we
did is copy the number into the extension buffer, if other matches
are now possible, there would thus be a timeout before the call went
through. We now complete redialed calls immediaetly in all cases.
Resolves: #1483
The bridge play and record APIs were forcing the Announcer/Recorder channel
to slin8 which meant that if you played or recorded audio with a sample
rate > 8K, it was downsampled to 8K limiting the bandwidth.
* The /bridges/play REST APIs have a new "announcer_format" parameter that
allows the caller to explicitly set the format on the "Announcer" channel
through which the audio is played into the bridge. If not specified, the
default depends on how many channels are currently in the bridge. If
a single channel is in the bridge, then the Announcer channel's format
will be set to the same as that channel's. If multiple channels are in the
bridge, the channels will be scanned to find the one with the highest
sample rate and the Announcer channel's format will be set to the slin
format that has an equal to or greater than sample rate.
* The /bridges/record REST API has a new "recorder_format" parameter that
allows the caller to explicitly set the format on the "Recorder" channel
from which audio is retrieved to write to the file. If not specified,
the Recorder channel's format will be set to the format that was requested
to save the audio in.
Resolves: #1479
DeveloperNote: The ARI /bridges/play and /bridges/record REST APIs have new
parameters that allow the caller to specify the format to be used on the
"Announcer" and "Recorder" channels respecitvely.
This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation.
The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header,
but that option had no effect as it was not implemented by res_pjsip_geolocation.
If the `location_source` configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation).
This commits adds already documented functionality.
This change moves observer invocation from the use of
a threadpool to a taskpool. The taskpool options have also
been adjusted to ensure that at least one taskprocessor
remains available at all times.
In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
AST_RTP_INSTANCE_RTCP_MUX is set.
Resolves: #1474
When handling SIP transfers via ARI, the `referred_by` field in
`transfer_ari_state` may be null, since SIP REFER requests are not
required to include a `Referred-By` header. Without this check, a null
value caused the transfer to fail and triggered a NOTIFY with a 500
Internal Server Error.
* Fixed an issue in webchan_write() where we weren't detecting equivalent
codecs properly.
* Added the "p" dialstring option that puts the channel driver in
"passthrough" mode where it will not attempt to re-frame or re-time
media coming in over the websocket from the remote app. This can be used
for any codec but MUST be used for codecs that use packet headers or whose
data stream can't be broken up on arbitrary byte boundaries. In this case,
the remote app is fully responsible for correctly framing and timing media
sent to Asterisk and the MEDIA text commands that could be sent over the
websocket are disabled. Currently, passthrough mode is automatically set
for the opus, speex and g729 codecs.
* Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
ensure proper translation paths are set up when switching between native
frames and slin silence frames. This fixes an issue with codec errors
when transcode_via_sln=yes.
Resolves: #1462
Add NULL check for word_list before calling word_in_list()
Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel()
Resolves: #1425
Add a dialplan function that can be used to get/set properties of
DAHDI channels (as opposed to Asterisk channels). This exposes
properties that were not previously available, allowing for certain
operations to now be performed in the dialplan.
Resolves: #1455
UserNote: The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
This change introduces a new API called taskpool. This is a pool
of taskprocessors. It provides the following functionality:
1. Task pushing to a pool of taskprocessors
2. Synchronous tasks
3. Serializers for execution ordering of tasks
4. Growing/shrinking of number of taskprocessors in pool
This functionality already exists through the combination of
threadpool+taskprocessors but through investigating I determined
that this carries substantial overhead for short to medium duration
tasks. The threadpool uses a single queue of work, and for management
of threads it involves additional tasks.
I wrote taskpool to eliminate the extra overhead and management
as much as possible. Instead of a single queue of work each
taskprocessor has its own queue and at push time a selector chooses
the taskprocessor to queue the task to. Each taskprocessor also
has its own thread like normal. This spreads out the tasks immediately
and reduces contention on shared resources.
Using the included efficiency tests the number of tasks that can be
executed per second in a taskpool is 6-12 times more than an equivalent
threadpool+taskprocessor setup.
Stasis has been moved over to using this new API as it is a heavy consumer
of threadpool+taskprocessors and produces a lot of tasks.
UpgradeNote: The threadpool_* options in stasis.conf have now been deprecated
though they continue to be read and used. They have been replaced with taskpool
options that give greater control over the underlying taskpool used for stasis.
DeveloperNote: The taskpool API has been added for common usage of a
pool of taskprocessors. It is suggested to use this API instead of the
threadpool+taskprocessor approach.
Added a check to outbound_websocket_apply() that makes sure an outbound
websocket config object in ari.conf has a websocket_client_id parameter.
Resolves: #1457
get_token can return NULL, but process_token uses this result without
checking for NULL; as elsewhere, check for a NULL result to avoid
possible NULL dereference.
Resolves: #1419
In a previous commit, a change was made to
ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample
rates. This ended up returning an invalid payload int for comfort noise.
A check has been added that returns early if the payload is in fact
supposed to be comfort noise.
Fixes: #1340
"dialplan eval function" has been using a dummy channel for function
evaluation, much like many of the unit tests. However, sometimes, this
can cause issues for functions that are not expecting dummy channels.
As an example, ast_channel_tech(chan) is NULL on such channels, and
ast_channel_tech(chan)->type consequently results in a NULL dereference.
Normally, functions do not worry about this since channels executing
dialplan aren't dummy channels.
While some functions are better about checking for these sorts of edge
cases, use a real channel with a dummy technology to make this CLI
command inherently safe for any dialplan function that could be evaluated
from the CLI.
Resolves: #1434
In many asterisk-based systems, the pause reason is used to separate
pauses by type,and logically, changing the reason defines two intervals
that should be accounted for separately. The introduction of a new
option allows me to separate the intervals of operator inactivity in
the log by the event of unpausing.
UserNote: Add new global option 'log_unpause_on_reason_change' that
is default disabled. When enabled cause addition of UNPAUSE event on
every re-PAUSE with reason changed.
The functions WaitForNoise() and WaitForSilence() use the time()
functions to calculate elapsed time, which causes the timer to fire on
a whole second boundary, and the actual function execution time to fire
the timer may be 1 second less than expected. This fix replaces time()
with ast_tvnow().
Fixes: #1401
Currently, the 'd' option will play dial tone while waiting
for digits. Allow it to accept an argument for any tone from
indications.conf.
Resolves: #1396
UserNote: The tone used while waiting for digits in WaitExten
can now be overridden by specifying an argument for the 'd'
option.
One of the problems with TONE_DETECT as it was originally written
is that if a tone is detected multiple times, it can trigger
the redirect logic multiple times as well. For example, if we
do an async goto in the dialplan after detecting a tone, because
the detector is still active until explicitly disabled, if we
detect the tone again, we will branch again and start executing
that dialplan a second time. This is rarely ever desired behavior,
and can happen if the detector is not removed quickly enough.
Add a new option, 'e', which automatically disables the detector
once the desired number of matches have been heard. This eliminates
the potential race condition where previously the detector would
need to be disabled immediately, but doing so quickly enough
was not guaranteed. This also allows match criteria to be retained
longer if needed, so the detector does not need to be destroyed
prematurely.
Resolves: #1390
UserNote: The 'e' option for TONE_DETECT now allows detection to
be disabled automatically once the desired number of matches have
been fulfilled, which can help prevent race conditions in the
dialplan, since TONE_DETECT does not need to be disabled after
a hit.
Numerically comparing that the current queue position is less than
last_pos_said can only be done after at least one announcement has been
made, otherwise last_pos_said is at the default (0).
Fixes: #1386
In the highly-unlikely event that get_authorization_hdr() couldn't find an
Authorization header in a request, trying to get the digest algorithm
would cauase a SEGV. We now check that we have an auth header that matches
the realm before trying to get the algorithm from it.
Resolves: #GHSA-64qc-9x89-rx5j
This patch resolves two issues in Sorcery objectset handling with multiple
backends:
1. Prevent duplicate objects:
When an object exists in more than one backend (e.g., a contact in both
'astdb' and 'realtime'), the objectset previously returned multiple instances
of the same logical object. This caused logic failures in components like the
PJSIP registrar, where duplicate contact entries led to overcounting and
incorrect deletions, when max_contacts=1 and remove_existing=yes.
This patch ensures only one instance of an object with a given key is added
to the objectset, avoiding these duplicate-related side effects.
2. Ensure missing objects are created:
When using multiple writable backends, a temporary backend failure can lead
to objects missing permanently from that backend.
Currently, .update() silently fails if the object is not present,
and no .create() is attempted.
This results in inconsistent state across backends (e.g. astdb vs. realtime).
This patch introduces a new global option in sorcery.conf:
[general]
update_or_create_on_update_miss = yes|no
Default: no (preserves existing behavior).
When enabled: if .update() fails with no data found, .create() is attempted
in that backend. This ensures that objects missing due to temporary backend
outages are re-synchronized once the backend is available again.
Added a new CLI command:
sorcery show settings
Displays global Sorcery settings, including the current value of
update_or_create_on_update_miss.
Updated tests to validate both flag enabled/disabled behavior.
Fixes: #1289
UserNote: Users relying on Sorcery multiple writable backends configurations
(e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
in sorcery.conf to ensure missing objects are recreated after temporary backend
failures. Default behavior remains unchanged unless explicitly enabled.
If Caller ID is disabled for an FXS port, then we should not send any
Caller ID spill on the line, as we have no Caller ID information that
we can/should be sending.
Resolves: #1394
It is possible to modify the dialmode setting in the chan_dahdi/sig_analog
private using the CHANNEL function, to modify it during calls. However,
it was not being reset between calls, meaning that if, for example, tone
dialing was disabled, it would never work again unless explicitly enabled.
This fixes the setting by pairing it with a "perm" version of the setting,
as a few other features have, so that it can be reset to the permanent
setting between calls. The documentation is also clarified to explain
the interaction of this setting and the digitdetect setting more clearly.
Resolves: #1378
* Added a new option to the WebSocket dial string to capture the additional
URI parameters.
* Added a new API ast_uri_verify_encoded() that verifies that a string
either doesn't need URI encoding or that it has already been encoded.
* Added a new API ast_websocket_client_add_uri_params() to add the params
to the client websocket session.
* Added XML documentation that will show up with `core show application Dial`
that shows how to use it.
Resolves: #1352
UserNote: A new WebSocket channel driver option `v` has been added to the
Dial application that allows you to specify additional URI parameters on
outgoing connections. Run `core show application Dial` from the Asterisk CLI
to see how to use it.
ast_websocket_read() receives data into a fixed 64K buffer then continually
reallocates a final buffer that, after all continuation frames have been
received, is the exact length of the data received and returns that to the
caller. process_text_message() in chan_websocket was attempting to set a
NULL terminator on the received payload assuming the payload buffer it
received was the large 64K buffer. The assumption was incorrect so when it
tried to set a NULL terminator on the payload, it could, depending on the
state of the heap at the time, cause heap corruption.
process_text_message() now allocates its own payload_len + 1 sized buffer,
copies the payload received from ast_websocket_read() into it then NULL
terminates it prevent the possibility of the overrun and corruption.
Resolves: #1384
Adds an ARI command to send a progress indication to a channel.
DeveloperNote: A new ARI endpoint is available at `/channels/{channelId}/progress` to indicate progress to a channel.
The debug logging during DSP processing has always been kind
of overwhelming and annoying to troubleshoot. Simplify and
improve the logging in a few ways to aid DSP debugging:
* If we had a DSP hit, don't also emit the previous debug message that
was always logged. It is duplicated by the hit message, so this can
reduce the number of debug messages during detection by 50%.
* Include the hit count and required number of hits in the message so
on partial detections can be more easily troubleshot.
* Use debug level 9 for hits instead of 10, so we can focus on hits
without all the noise from the per-frame debug message.
* 1-index the hit count in the debug messages. On the first hit, it
currently logs '0', just as when we are not detecting anything,
which can be confusing.
Resolves: #1375
After an asterisk restart, the deletion of ARI Devicestates didn't
return error, but the devicestate was not deleted.
Found a typo on populate_cache function that created wrong cache for
device states.
This bug caused wrong assumption that devicestate didn't exist,
since it was not in cache, so deletion didn't returned error.
Fixes: #1327
Add an option for ChanSpy and ExtenSpy to not answer the channel
automatically. Most applications that auto-answer by default
already have an option to disable this behavior if unwanted.
Resolves: #1358
UserNote: ChanSpy and ExtenSpy can now be configured to not
automatically answer the channel by using the 'N' option.
If you do a `core show application Dial`, you'll see it's kind of a mess.
Indents are wrong is some places, examples are printed in black which makes
them invisible on most terminals, and the lack of line breaks in some cases
makes it hard to follow.
* Fixed the rendering of examples so they are indented properly and changed
the color so they can be seen.
* There is now a line break before each option.
* Options are now printed on their own line with all option content indented
below them.
Example from Dial before fixes:
```
Example: Dial 555-1212 on first available channel in group 1, searching
from highest to lowest
Example: Ringing FXS channel 4 with ring cadence 2
Example: Dial 555-1212 on channel 3 and require answer confirmation
...
O([mode]):
mode - With <mode> either not specified or set to '1', the originator
hanging up will cause the phone to ring back immediately.
- With <mode> set to '2', when the operator flashes the trunk, it will ring
their phone back.
Enables *operator services* mode. This option only works when bridging a DAHDI
channel to another DAHDI channel only. If specified on non-DAHDI interfaces, it
will be ignored. When the destination answers (presumably an operator services
station), the originator no longer has control of their line. They may hang up,
but the switch will not release their line until the destination party (the
operator) hangs up.
p: This option enables screening mode. This is basically Privacy mode
without memory.
```
After:
```
Example: Dial 555-1212 on first available channel in group 1, searching
from highest to lowest
same => n,Dial(DAHDI/g1/5551212)
Example: Ringing FXS channel 4 with ring cadence 2
same => n,Dial(DAHDI/4r2)
Example: Dial 555-1212 on channel 3 and require answer confirmation
same => n,Dial(DAHDI/3c/5551212)
...
O([mode]):
mode - With <mode> either not specified or set to '1', the originator
hanging up will cause the phone to ring back immediately.
With <mode> set to '2', when the operator flashes the trunk, it will
ring their phone back.
Enables *operator services* mode. This option only works when bridging
a DAHDI channel to another DAHDI channel only. If specified on
non-DAHDI interfaces, it will be ignored. When the destination answers
(presumably an operator services station), the originator no longer has
control of their line. They may hang up, but the switch will not
release their line until the destination party (the operator) hangs up.
p:
This option enables screening mode. This is basically Privacy mode
without memory.
```
There are still things we can do to make this more readable but this is a
start.
This can be reproduced under certain curcomstences.
For example: call app.playback from lua with invalid data: app.playback({}).
pbx_lua.c will try to get data for this playback using lua_tostring function.
This function returs NULL for everything but strings and numbers.
Then, it calls term_color with NULL data.
term_color function can call(if we don't use vt100 compat term)
ast_copy_string with NULL inbuf which cause segfault. bt example:
ast_copy_string (size=8192, src=0x0, dst=0x7fe44b4be8b0)
at /usr/src/asterisk/asterisk-20.11.0/include/asterisk/strings.h:412
Resolves: https://github.com/asterisk/asterisk/issues/1363
If the BRIDGE_NOANSWER variable is set on a channel, it is not supposed
to answer when another channel bridges to it using Bridge(), and this is
checked when ast_bridge_call* is called. However, another path exists
(bridge_exec -> ast_bridge_add_channel) where this variable was not
checked and channels would be answered. We now check the variable there.
Resolves: #401Resolves: #1364
