This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
which when set causes the ODBC variant of app_voicemail to leave
the message and greeting audio files on disk and only store the
message metadata in the database. This option came from a concern
that the database could grow to large and cause remote access
and/or replication to become slow. In a clustering situation
with this option, all asterisk instances would share the same
database for the metadata and either use a shared filesystem
or other filesystem replication service much more suitable
for synchronizing files.
The changes to app_voicemail to implement this feature were actually
quite small but due to the complexity of the module, the actual
source code changes were greater. They fall into the following
categories:
* Tracing. The module is so complex that it was impossible to
figure out the path taken for various scenarios without the addition
of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
code that's not related to the functional change. Making this worse
was the fact that many "if" statements in this module didn't use
braces. Since the tracing macros add multiple statements, many "if"
statements had to be converted to use braces.
* Excessive use of PATH_MAX. Previous maintainers of this module
used PATH_MAX to allocate character arrays for filesystem paths
and SQL statements as though they cost nothing. In fact, PATH_MAX
is defined as 4096 bytes! Some functions had (and still have)
multiples of these. One function has 7. Given that the vast
majority of installations use the default spool directory path
`/var/spool/asterisk/voicemail`, the actual path length is usually
less than 80 bytes. That's over 4000 bytes wasted. It was the
same for SQL statement buffers. A 4K buffer for statement that
only needed 60 bytes. All of these PATH_MAX allocations in the
ODBC related code were changed to dynamically allocated buffers.
The rest will have to be addressed separately.
* Bug fixes. During the development of this feature, several
pre-existing ODBC related bugs were discovered and fixed. They
had to do with leaving orphaned files on disk, not preserving
original message ids when moving messages between folders,
not honoring the "formats" config parameter in certain circumstances,
etc.
UserNote: This commit adds a new voicemail.conf option
'odbc_audio_on_disk' which when set causes the ODBC variant of
app_voicemail_odbc to leave the message and greeting audio files
on disk and only store the message metadata in the database.
Much more information can be found in the voicemail.conf.sample
file.
From the gdb information, we can see that while iterating over bridge->channels, the ast_bridge_channel reference count is 0, indicating it has already been destroyed.Additionally, when ast_bridge_channel is removed from bridge->channels, the bridge is first locked. Therefore, locking the bridge before iterating over bridge->channels can resolve the race condition.
Resolves: https://github.com/asterisk/asterisk/issues/768
Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event
in the queue log if the Caller ID is restricted.
Resolves: #765
UpgradeNote: Add a new column to the queues table:
queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
to control whether the Restricted Caller ID will be stored in the queue log.
UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
will be stored in the queue log.
If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.
The current width for "extension" is 20 and "device state id" is 20, which is too small.
The "extension" field contains "ext"@"context", so 20 characters is not enough.
The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough.
Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.
Resolves: #770
UserNote: The fields width of "core show hints" were increased.
The width of "extension" field to 30 characters and
the width of the "device state id" field to 60 characters.
Various SIP headers permit a URI to be prefaced with a `display-name`
production that can include characters (like commas and parentheses)
that are problematic for Asterisk's dialplan parser and, specifically
in the case of this patch, the PJSIP_PARSE_URI function.
This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that
behaves identically to `PJSIP_PARSE_URI` except that the first
argument is now a variable name and not a literal URI.
Fixes#756
Since Asterisk 19 it is possible to cache recorded files into another
directory [1] [2].
Show configured location of cache dir in CLI's core show settings.
[1] ASTERISK-29143
[2] b08427134f
Two functions are deprecated as of libxml2 2.12:
* xmlSubstituteEntitiesDefault
* xmlParseMemory
So we update those with supported API.
Additionally, `res_calendar_caldav` has been updated to use libxml2's
xmlreader API instead of the SAX2 API which has always felt a little
hacky (see deleted comment block in `res_calendar_caldav.c`).
The xmlreader API has been around since libxml2 2.5.0 which was
released in 2003.
Fixes#725
Missing or corrupt cdr_pgsql.conf configuration file can cause the
second attempt to load the PostgreSQL CDR module to crash Asterisk via
the Command Line Interface because a null CLI command is registered on
the first failed attempt to load the module.
Resolves: #736
Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.
Asterisk currently treats RFC2833 Digits as a single rtp payload type
with a fixed bitrate of 8K. This change would expand that to 8, 16,
24 and 32K.
This requires checking the offered rtp types for any of these bitrates
and then adding an offer for each (if configured for RFC2833.) DTMF
generation must also be changed in order to look at the current outbound
codec in order to generate appropriately timed rtp.
For cases where no outgoing audio has yet been sent prior to digit
generation, Asterisk now has a concept of a 'preferred' codec based on
offer order.
On inbound calls Asterisk will mimic the payload types of the RFC2833
digits.
On outbound calls Asterisk will choose the next free payload types starting
with 101.
UserNote: No change in configuration is required in order to enable this
feature. Endpoints configured to use RFC2833 will automatically have this
enabled. If the endpoint does not support this, it should not include it in
the SDP offer/response.
Resolves: #699
Send "RELOADING=1" instead of "RELOAD=1" to follow the format
expected by systemd (see sd_notify(3) man page).
Do not send STOPPING=1 in remote console mode:
attempting to execute "asterisk -rx" by the main process leads to
a warning if NotifyAccess=main (the default) or to a forced termination
if NotifyAccess=all.
Include signal.h to avoid the following build failure with uclibc-ng
raised since
2694792e13:
stasis/control.c: In function 'exec_command_on_condition':
stasis/control.c:313:3: warning: implicit declaration of function 'pthread_kill'; did you mean 'pthread_yield'? [-Wimplicit-function-declaration]
313 | pthread_kill(control->control_thread, SIGURG);
| ^~~~~~~~~~~~
| pthread_yield
stasis/control.c:313:41: error: 'SIGURG' undeclared (first use in this function)
313 | pthread_kill(control->control_thread, SIGURG);
| ^~~~~~
cherry-pick-to: 18
cherry-pick-to: 20
cherry-pick-to: 21
Fixes: #729
Currently, reloading res_pjsip will cause logging
to be disabled. This is because logging can also
be controlled via the debug option in pjsip.conf
and this defaults to "no".
To improve this, logging is no longer disabled on
reloads if logging had not been previously
enabled using the debug option from the config.
This ensures that logging enabled from the CLI
will persist through a reload.
ASTERISK-29912 #close
Resolves: #246
UserNote: Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
Some of the money announcements can be off by one cent,
due to the use of floating point in the money calculations,
which is bad for obvious reasons.
This replaces floating point with simple string parsing
to ensure the cents value is converted accurately.
Resolves: #525
Because of the (often recursive) nature of module dependencies in
Asterisk, hot swapping a module on the fly is cumbersome if a module
is depended on by other modules. Currently, dependencies must be
popped manually by unloading dependents, unloading the module of
interest, and then loading modules again in reverse order.
To make this easier, the ability to do this recursively in certain
circumstances has been added, as an optional extension to the
"module refresh" command. If requested, Asterisk will check if a module
that has a positive usecount could be unloaded safely if anything
recursively dependent on it were unloaded. If so, it will go ahead
and unload all these modules and load them back again. This makes
hot swapping modules that provide dependencies much easier.
Resolves: #474
UserNote: In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation.
Fixes#710
If the hostname field of the ast_tcptls_session_args structure is
set (which it is for websocket client connections), that hostname
will now automatically be used in an SNI TLS extension in the client
hello.
Resolves: #713
UserNote: Secure websocket client connections now send SNI in
the TLS client hello.
* Fixed possible memory leak in tn_config:tn_get_etn() where we
weren't releasing etn if tn or eprofile were null.
* We now canonicalize TNs before using them for lookups or adding
them to Identity headers.
* Fixed a typo in stir_shaken.conf.sample.
Resolves: #716
Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must
always be included in buildopts.h instead of only when
ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise.
Resolves: #719
Commit f2f397c1a8 previously
made it possible to send Caller ID parameters to FXS stations
which, prior to that, could not be sent.
This change is complementary in that we now handle receiving
all these parameters on FXO lines and provide these up to
the dialplan, via chan_dahdi. In particular:
* If a redirecting reason is provided, the channel's redirecting
reason is set. No redirecting number is set, since there is
no parameter for this in the Caller ID protocol, but the reason
can be checked to determine if and why a call was forwarded.
* If the Call Qualifier parameter is received, the Call Qualifier
variable is set.
* Some comments have been added to explain why some of the code
is the way it is, to assist other people looking at it.
With this change, Asterisk's Caller ID implementation is now
reasonably complete for both FXS and FXO operation.
Resolves: #681
If you're tracing a large function that may call another function
multiple times in different circumstances, it can be difficult to
see from the trace output exactly which location that function
was called from. There's no good way to automatically determine
the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT
simply print out a trace line before and after the call.
The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is
that SCOPE_CALL ignores the function's return value (if any) where
SCOPE_CALL_WITH_RESULT allows you to specify the type of the
function's return value so it can be assigned to a variable.
SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT
and the "int" return type.
rtp_engine.c and stun.c were calling ast_register_cleanup which
is skipped if any loadable module can't be cleanly unloaded
when asterisk shuts down. Since this will always be the case,
their cleanup functions never get run. In a practical sense
this makes no difference since asterisk is shutting down but if
you're in development mode and trying to use the leak sanitizer,
the leaks from both of those modules clutter up the output.
Emit a warning if REDIRECTING(reason) is set to an invalid
reason, consistent with what happens when
REDIRECTING(orig-reason) is set to an invalid reason.
Resolves: #683
* Add an AMI action to correspond to the "dahdi show status"
command, allowing span information to be retrieved via AMI.
* Show span number and sig type in "dahdi show channels".
Resolves: #673
Add a new identify_by option to res_pjsip_endpoint_identifier_ip
called 'transport' this matches endpoints based on the bound
ip address (local) instead of the 'ip' option, which matches on
the source ip address (remote).
UserNote: set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
* OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use
X509_get_pubkey. The difference is that X509_get_pubkey requires
the caller to free the EVP_PKEY themselves so we now let
RAII_VAR do that.
* OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now
wrap it in an ao2 object.
* OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all
the certs from an X509_STORE and there's no easy way to polyfill
it so the CLI commands that list profiles will show a "not
supported" message instead of listing the certs in a store.
Resolves: #676
There were a few references in the embedded documentation XML
where the case didn't match or where the referenced app or function
simply didn't exist any more. These were causing 404 responses
in docs.asterisk.org.
Add ability to match against PJSIP request URI.
UserNote: this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
Commit 729cb1d390 added logic to retry
opening DAHDI channels on "dahdi restart" if they failed initially,
up to 1,000 times in a loop, to address cases where the channel was
still in use. However, this retry loop does not use the actual error,
which means chan_dahdi will also retry opening nonexistent channels
1,000 times per channel, causing a flood of unnecessary warning logs
for an operation that will never succeed, with tens or hundreds of
thousands of open attempts being made.
The original patch would have been more targeted if it only retried
on the specific relevant error (likely EBUSY, although it's hard to
say since the original issue is no longer available).
To avoid the problem above while avoiding the possibility of breakage,
this skips the retry logic if the error is ENXIO (No such device or
address), since this will never succeed.
Resolves: #669
This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary.
Fixes#657
There was functionality in chan_sip to get REFER headers, with GET_TRANSFERRER_DATA variable. This commit implements the same functionality in pjsip, to ease transfer from chan_sip to pjsip.
Fixes: #579
UserNote: the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
SQLAlchemy 2.0 changed the way that commits/rollbacks are handled
causing the final `UPDATE` to our `alembic_version_<whatever>` tables
to be rolled back instead of committed.
We now use one connection to determine which
`alembic_version_<whatever>` table to use and another to run the
actual migrations. This prevents the erroneous rollback.
This change is compatible with both SQLAlchemy 1.4 and 2.0.
manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
The action originate does not has the ability to run an subroutine at initial channel, like the Aplication Originate. This update give this ability for de action originate too.
For example, we can run a routine via Gosub on the channel to request an automatic answer, so the caller does not need to accept the call when using the originate command via manager, making the operation more efficient.
UserNote: When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
