Currently, if chan_websocket receives an un supported frame like comfort noise it will exit the websocket. The proposed change is to tolerate the other frames by not sending them down the websocket but instead just ignoring them.
Resolves: #1587
Calling Reload() without any arguments is supposed to reload
everything (equivalent to a 'core reload'), but actually does
nothing. This is because it was calling ast_module_reload with
an empty string, and the argument needs to explicitly be NULL.
Resolves: #1597
When running "dialplan reload", the number of contexts reported
is initially wrong, as it is the old context count. Running
"dialplan reload" a second time returns the correct number of
contexts that are loaded. This can confuse users into thinking
that the reload didn't work successfully the first time.
This counter is currently only incremented when iterating the
old contexts prior to the context merge; at the very end, get
the current number of elements in the context hash table and
report that instead. This way, the count is correct immediately
whenever a reload occurs.
Resolves: #1599
Convenience wrappers for showing modules at various support levels.
* module-list-core
* module-list-extended
* module-list-deprecated
Resolves: #1572
UserNote: Try "make module-list-deprecated" to see what modules
are on their way out the door.
Commit a46d5f9b76 removed the deprecated
'e' option to ResetCDR; this now causes DISA() to emit a warning
if attempting to call ResetCDR() with the deprecated option (in
all cases except when the no answer option is provided). Rewrite
the code to do this the current way.
Resolves: #1592
After p->chan = NULL, ast still points to the valid channel object,
using ast safely accesses the channel's DIALSTATUS variable before it's fully destroyed
Resolves: #1590
* Fixed an issue with tarball-coredumps when asterisk was invoked without an
absolute path.
* Fixed an issue with gdb itself segfaulting when trying to get symbols from
separate debuginfo files. The command line arguments needed to be altered
such that the gdbinit files is loaded before anything else but the
`dump-asterisk` command is run after full initialization.
In the embedded gdbinit script:
* The extract_string_symbol function needed a `char *` cast to work properly.
* The s_strip function needed to be updated to continue to work with the
cpp_map_name_id channel storage backend.
* A new function was added to dump the channels when cpp_map_name_id was
used.
* The Channel object was updated to account for the new channel storage
backends
* The show_locks function was refactored to work correctly.
The 's' (skip) option delays MixMonitor recording until the specified number of seconds
(can be fractional) have elapsed since MixMonitor was invoked.
No audio is written to the recording file during this time. If the call ends before this
period, no audio will be saved. This is useful for avoiding early audio such as
announcements, ringback tones, or other non-essential sounds.
UserNote: This change introduces a new 's(<seconds>)' (skip) option to the MixMonitor
application. Example:
MixMonitor(${UNIQUEID}.wav,s(3))
This skips recording for the first 3 seconds before writing audio to the file.
Existing MixMonitor behavior remains unchanged when the 's' option is not used.
Only make announcements to head caller if announce_to_first_user is true
Fixes: #1568
UserNote: When announce_to_first_user is false, no announcements are played to the head caller
After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.
Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.
The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.
Resolves: #1578
Also cleaned up a few unused #if blocks, and started sending a few ERROR
events back to the apps.
Resolves: #1574
DeveloperNote: Apps can now send a `MARK_MEDIA` command with an optional
`correlation_id` parameter to chan_websocket which will be placed in the
media frame queue. When that frame is dequeued after all intervening media
has been played to the core, chan_websocket will send a
`MEDIA_MARK_PROCESSED` event to the app with the same correlation_id
(if any).
With recent enhancements to chan_websocket, the original plain-text
implementation of control messages and events is now too limiting. We
probably should have used JSON initially but better late than never. Going
forward, enhancements that require control message or event changes will
only be done to the JSON variants and the plain-text variants are now
deprecated but not yet removed.
* Added the chan_websocket.conf config file that allows setting which control
message format to use globally: "json" or "plain-text". "plain-text" is the
default for now to preserve existing behavior.
* Added a dialstring option `f(json|plain-text)` to allow the format to be
overridden on a call-by-call basis. Again, 'plain-text' is the default for
now to preserve existing behavior.
The JSON for commands sent by the app to Asterisk must be...
`{ "command": "<command>" ... }` where `<command>` is one of `ANSWER`, `HANGUP`,
`START_MEDIA_BUFFERING`, etc. The `STOP_MEDIA_BUFFERING` command takes an
additional, optional parameter to be returned in the corresponding
`MEDIA_BUFFERING_COMPLETED` event:
`{ "command": "STOP_MEDIA_BUFFERING", "correlation_id": "<correlation id>" }`.
The JSON for events sent from Asterisk to the app will be...
`{ "event": "<event>", "channel_id": "<channel_id>" ... }`.
The `MEDIA_START` event will now look like...
```
{
"event": "MEDIA_START",
"connection_id": "media_connection1",
"channel": "WebSocket/media_connection1/0x5140001a0040",
"channel_id": "1761245643.1",
"format": "ulaw",
"optimal_frame_size": 160,
"ptime": 20,
"channel_variables": {
"DIALEDPEERNUMBER": "media_connection1/c(ulaw)",
"MEDIA_WEBSOCKET_CONNECTION_ID": "media_connection1",
"MEDIA_WEBSOCKET_OPTIMAL_FRAME_SIZE": "160"
}
}
```
Note the addition of the channel variables which can't be supported
with the plain-text formatting.
The documentation will be updated with the exact formats for all commands
and events.
Resolves: #1546Resolves: #1563
DeveloperNote: The chan_websocket plain-text control and event messages are now
deprecated (but remain the default) in favor of JSON formatted messages.
See https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket for
more information.
DeveloperNote: A "transport_data" parameter has been added to the
channels/externalMedia ARI endpoint which, for websocket, allows the caller
to specify parameters to be added to the dialstring for the channel. For
instance, `"transport_data": "f(json)"`.
The following options have been added to the menuselect "Compiler Flags"
section...
CODE_COVERAGE: The ability to enable code coverage via the `--enable-coverage`
configure flag has existed for many years but changing it requires
re-running ./configure which is painfully slow. With this commit, you can
now enable and disable it via menuselect. Setting this option adds the
`-ftest-coverage` and `-fprofile-arcs` flags on the gcc and ld command lines.
It also sets DONT_OPTIMIZE. Note: If you use the `--enable-coverage` configure
flag, you can't turn it off via menuselect so choose one method and stick to
it.
KEEP_FRAME_POINTERS: This option sets `-fno-omit-frame-pointers` on the gcc
command line which can facilitate debugging with 'gdb' and tracing with 'perf'.
Unlike CODE_COVERAGE, this option doesn't depend on optimization being
disabled. It does however conflict with COMPILE_DOUBLE.
Operations on non-blocking sockets may return a resource temporarily unavailable error (EAGAIN or EWOULDBLOCK). This is not a fatal error but a normal condition indicating that the operation would block.
This patch corrects the handling of this case. Instead of incorrectly treating it as a reason to terminate the connection, the code now waits for data to arrive on the socket.
* Using `==` with the POSIX sh `test` utility is UB.
* Switch back to using globs instead of using `$(find … | sort)`.
* Fix a missing redirect when checking for the OS type.
Resolves: #1554
Also...
* Refactored the verification datastore process so instead of having
a separate channel datastore for each verification result, there's only
one channel datastore with a vector of results.
* Refactored some log messages to include channel name and removed
some that would be redundant if a memory allocation failed.
Resolves: #781
UserNote: The STIR_SHAKEN_ATTESTATION dialplan function has been added
which will allow suppressing attestation on a call-by-call basis
regardless of the profile attached to the outgoing endpoint.
The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake().
In this case, a timeout mechanism should be set for the SSL/TLS handshake process to prevent indefinite waiting during the SSL handshake.
Resolves: #1559
Commit b333ee3b introduced a fix to chan_pjsip that addressed RTP issues with
blind transfers and some SBCs. Unfortunately, the fix broke some WebRTC
clients that are sensitive to SSRC changes and non-monotonic timestamps so
the fix is now disabled for endpoints with the "bundle" parameter set to true.
Resolves: #1535
This change moves the PJSIP module from the threadpool API
to the taskpool API. PJSIP-specific implementations for
task usage have been removed and replaced with calls to
the optimized taskpool implementations instead. The need
for a pool of serializers has also been removed as
taskpool inherently provides this. The default settings
have also been changed to be more realistic for common
usage.
UpgradeNote: The threadpool_* options in pjsip.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 PJSIP. An alembic
upgrade script has been added to add these options to realtime
as well.
Re-enabled "TTY=9" which was erroneously disabled as part of a recent
security fix and removed another logging "fix" that was added.
Also added a sort to the "find" that enumerates the scripts to be sourced so
they're sourced in the correct order.
Resolves: #1539
Commit 43bf8a4ded introduced endian
dependend byte-swapping code in include/asterisk/utils.h, where the
endianness was detected using the __BYTE_ORDER macro. This macro
lives in endian.h, which on Linux is included implicitely (by the
network-related headers, I think), but on FreeBSD the headers are
laid out differently and we do not get __BYTE_ORDER the implicit way.
Instead, this makes the usage of endian.h explicit by including it
where we need it, and switches the BYTE_ORDER/*ENDIAN macros to the
POSIX-defined ones (see
https://pubs.opengroup.org/onlinepubs/9799919799/basedefs/endian.h.html
for standard compliance). Additionally, this adds a compile-time check
for the endianness-logic: compilation will fail if neither big nor
little endian can be detected.
Fixes: #1536
When macro was removed in Asterisk 21, the parameter documentation in
code was not updated to reflect the correct numerization for gosub. It
still stated that it was the seventh parameter, but got shifted to the
sixth due to the removal of macro. This has been updated to correctly
reflect the parameter order, and a note has been added to the XML that
states this was done after the initial commit.
Fixes: #1534
UpgradeNote: As part of Asterisk 21, macros were removed from Asterisk.
This resulted in argument order changing for the Queue dialplan
application since the macro argument was removed. Upgrade notice was
missed when this was done, so this upgrade note has been added to
provide a record of such and a notice to users who may have not upgraded
yet.
When publishing device state check the local cache for the
existing device state. If the new device state is unchanged
from the prior one, don't bother publishing the update. This
can reduce the work done by consumers of device state, such
as hints and app_queue, by not publishing a message to them.
These messages would most often occur with devices that are
seeing numerous simultaneous channels. The underlying device
state would remain as in use throughout, but an update would
be published as channels are created and hung up.
Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI
and AMI hangup and channel destroyed events, it can be helpful to know what
the actual channel technology code was if the call was unsuccessful.
For PJSIP, it's the SIP response code.
* A new "tech_hangupcause" field was added to the ast_channel structure along
with ast_channel_tech_hangupcause() and ast_channel_tech_hangupcause_set()
functions. It should only be set for off-nominal terminations.
* chan_pjsip was modified to set the tech hangup cause in the
chan_pjsip_hangup() and chan_pjsip_session_end() functions. This is a bit
tricky because these two functions aren't always called in the same order.
The channel that hangs up first will get chan_pjsip_session_end() called
first which will trigger the core to call chan_pjsip_hangup() on itself,
then call chan_pjsip_hangup() on the other channel. The other channel's
chan_pjsip_session_end() function will get called last. Unfortunately,
the other channel's HangupRequest events are sent before chan_pjsip has had a
chance to set the tech hangupcause code so the HangupRequest events for that
channel won't have the cause code set. The ChannelDestroyed and Hangup
events however will have the code set for both channels.
* A new "tech_cause" field was added to the ast_channel_snapshot_hangup
structure. This is a public structure so a bit of refactoring was needed to
preserve ABI compatibility.
* The ARI ChannelHangupRequest and ChannelDestroyed events were modified to
include the "tech_cause" parameter in the JSON for off-nominal terminations.
The parameter is suppressed for nominal termination.
* The AMI SoftHangupRequest, HangupRequest and Hangup events were modified to
include the "TechCause" parameter for off-nominal terminations. Like their ARI
counterparts, the parameter is suppressed for nominal termination.
DeveloperNote: A "tech_cause" parameter has been added to the
ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause"
parameter has been added to the HangupRequest, SoftHangupRequest and Hangup
AMI event messages. For chan_pjsip, these will be set to the last SIP
response status code for off-nominally terminated calls. The parameter is
suppressed for nominal termination.
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.
