Extend audiosocket messages with types 0x11 - 0x18 to create asterisk
frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 format, enabling the transmission of audio at a higher sample
rates. For audiosocket messages sent by Asterisk, the message kind is
determined by the format of the originating asterisk frame.
UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 audio. External applications using audiosocket may need to be
updated to support these message types if the audiosocket channel is
created with one of these audio formats.
Reading ${FAXOPT()} before a fax session is common in dialplans to check fax state.
Currently this logs an error even when no fax datastore exists, creating excessive noise.
Change these messages to ast_debug(3, …) so they appear only with debug enabled.
Resolves: #1509
When an endpoint is created in the core of Asterisk a subscription
was previously created alongside it to monitor any channels being
destroyed that were related to it. This was done by receiving all
channel snapshot updates for every channel and only reacting when
it was indicated that the channel was dead.
This change removes this logic and instead provides an API call
for directly removing a channel from an endpoint. This is called
when channels are destroyed. This operation is fast, so blocking
the calling thread for a short period of time doesn't have any
noticeable impact.
The app_queue module subscribes on a per-dialed agent basis to both
the bridge all and channel all topics to keep apprised of things going
on involving them. This subscription has associated state that must
be cleaned up when the subscription ends. This was done by setting
a default router callback that only had logic to handle the case
where the subscription ends. By using the default router callback
all filtering for the subscription was disabled, causing unrelated
messages to get published and handled by it.
This change makes it so that an explicit route is added for the
message type used for the message indicating the subscription has
ended and removes the default router callback. This allows message
filtering to occur on publishing reducing the messages to app_queue
to only those it is interested in.
The Dial() application does not propagate DNID fields, which is counter
to the behavior of the other Caller ID fields. This behavior is likely
intentional since the use of Dial theoretically suggests a new dialed
number, but document this caveat to inform users of it.
Resolves: #1519
Allow retrieving and setting the channel's ADSI capability from the
dialplan.
Resolves: #1514
UserNote: CHANNEL(adsicpe) can now be read or written to change
the channels' ADSI CPE capability setting.
Dial() already preserves the ADSI capability by copying it to the new
channel, but since Local channel pairs consist of two channels, we
also need to copy the capability to the second channel.
Resolves: #1517
As soon as SIP call may end with several Reason headers, we
want to make all of them available through the HAGUPCAUSE() function.
This implementation uses the same ao2 hash for cause codes storage
and adds a flag to make difference between last processed sip
message and content of reason headers.
UserNote: Added a new option to HANGUPCAUSE to access additional
information about hangup reason. Reason headers from pjsip
could be read using 'tech_extended' cause type.
It is customary to allow # to terminate digit collection immediately
when there would normally be a timeout. However, currently, users are
forced to wait for the timeout to expire when dialing numbers that
are prefixes of other valid matches, and there is no way to end the
timeout early. Customarily, # terminates the timeout, but at the moment,
this is just rejected unless there happens to be a matching extension
ending in #.
Allow # to terminate the timeout in cases where there is no dialplan
match. This ensures that the dialplan is always respected, but if a
valid extension has been dialed that happens to prefix other valid
matches, # can be used to dial it immediately.
Resolves: #1510
Add a function (DIGIT_SUM) which returns the digit sum of a number.
Resolves: #1499
UserNote: The DIGIT_SUM function can be used to return the digit sum of
a number.
Add a sorely needed option to set a timeout between digits, rather than
for receiving the entire number. This is needed if the number of digits
being sent is unknown by the receiver in advance. Previously, we had
to wait for the entire timer to expire.
Resolves: #1493
UserNote: The 't' option for ReceiveSF now allows for a timer since
the last digit received, in addition to the number-wide timeout.
Among the lower-quality voice codecs, some of the quality scores did
not make sense relative to each other.
For instance, quality-wise, G.729 > G.723 > PLC10.
However, current scores do not uphold these relationships.
Tweak the scores slightly to reflect more accurate relationships.
Resolves: #1501
When we retrieve a channel from a C++ map, we actually get back a wrapper
object that points to the channel then right after we retrieve it, we bump its
reference count. There's a tiny chance however that between those two
statements a delete and/or unref might happen which would cause the wrapper
object or the channel itself to become invalid resulting in a SEGV. To avoid
this we now perform a read lock on the driver around those statements.
Resolves: #1491
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.
Follow-on to #244 and #960 regarding how the ast_config_XXX APIs
handle template inheritance.
ast_config_text_file_save2() incorrectly suppressed variables if they
matched any ancestor template. This broke deep chains (dropping values
based on distant parents) and wide inheritance (ignoring last-wins order
across multiple parents).
The function now inspects the full template hierarchy to find the nearest
effective parent (last occurrence wins). Earlier inherited duplicates are
collapsed, explicit overrides are kept unless they exactly match the parent,
and PreserveEffectiveContext avoids writing redundant lines.
Resolves: #1451
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
There was no check in __rtp_sendto that prevented Asterisk from sending
RTP before DTLS had finished negotiating. This patch adds logic to do
so.
Fixes: #1260
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.
