* The autoconf-archive package contains macros useful for detecting C++
standard and testing other C++ capabilities but that package was never
included in the install_prereq script so many existing build environments
won't have it. Even if it is installed, older versions won't newer C++
standards and will actually cause an error if you try to test for that
version. To make it available for those environments, the
ax_cxx_compile_stdcxx.m4 macro has copied from the latest release of
autoconf-archive into the autoconf directory.
* A convenience wrapper(ast_cxx_check_std) around ax_cxx_compile_stdcxx was
also added so checking the standard version and setting the
asterisk-specific PBX_ variables becomes a one-liner:
`AST_CXX_CHECK_STD([std], [force_latest_std])`.
Calling that with a version of `17` for instance, will set PBX_CXX17
to 0 or 1 depending on whether the current c++ compiler supports stdc++17.
HAVE_CXX17 will also be 'defined" or not depending on the result.
* C++ compilers hardly ever default to the latest standard they support. g++
version 14 for instance supports up to C++23 but only uses C++17 by default.
If you want to use C++23, you have to add `-std=gnu++=23` to the g++
command line. If you set the second argument of AST_CXX_CHECK_STD to "yes",
the macro will automatically keep the highest `-std=gnu++` value that
worked and pass that to the Makefiles.
* The autoconf-archive package was added to install_prereq for future use.
* Updated configure.ac to use AST_CXX_CHECK_STD() to check for C++
versions 11, 14, 17, 20 and 23.
* Updated configure.ac to accept the `--enable-latest-cxx-std` option which
will set the second option to AST_CXX_CHECK_STD() to "yes". The default
is "no".
* ast_copy_string() in strings.h declares the 'sz' variable as volatile and
does an `sz--` on it later. C++20 no longer allows the `++` and `--`
increment and decrement operators to be used on variables declared as
volatile however so that was changed to `sz -= 1`.
(cherry picked from commit c193752ca4)
Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify
feature to mark a contact as available only if an OPTIONS request
returns a 2XX response. If the option is not specified or is false,
any response to the OPTIONS request marks the contact as available.
UserNote: The pjsip.conf AOR section now has a "qualify_2xx_only"
option that can be set so that only 2XX responses to OPTIONS requests
used to qualify a contact will mark the contact as available.
(cherry picked from commit 0b30f546ba)
The suppress_moh_on_sendonly endpoint option should have been
defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES
in the alembic script instead of OPT_YESNO_T and YESNO_VALUES.
Also updated contrib/ast-db-manage/README.md to indicate that
AST_BOOL_VALUES should always be used and provided an example.
Resolves: #995
Normally, when one party in a call sends Asterisk an SDP with
a "sendonly" or "inactive" attribute it means "hold" and causes
Asterisk to start playing MOH back to the other party. This can be
problematic if it happens at certain times, such as in a 183
Progress message, because the MOH will replace any early media you
may be playing to the calling party. If you set this option
to "yes" on an endpoint and the endpoint receives an SDP
with "sendonly" or "inactive", Asterisk will NOT play MOH back to
the other party.
Resolves: #979
UserNote: The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
(cherry picked from commit 98510d4c75)
This patch introduces a new identifier for channels: tenantid. It's
a stringfield on the channel that can be used for general purposes. It
will be inherited by other channels the same way that linkedid is.
You can set tenantid in a few ways. The first is to set it in the
dialplan with the Set and CHANNEL functions:
exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)
It can also be accessed via CHANNEL:
exten => example,2,NoOp(CHANNEL(tenantid))
Another method is to use the new tenantid option for pjsip endpoints in
pjsip.conf:
[my_endpoint]
type=endpoint
tenantid=My tenant ID
This is considered the best approach since you will be able to see the
tenant ID as early as the Newchannel event.
It can also be set using set_var in pjsip.conf on the endpoint like
setting other channel variable:
set_var=CHANNEL(tenantid)=My tenant ID
Note that set_var will not show tenant ID on the Newchannel event,
however.
Tenant ID has also been added to CDR. It's read-only and can be accessed
via CDR(tenantid). You can also get the tenant ID of the last channel
communicated with via CDR(peertenantid).
Tenant ID will also show up in CEL records if it has been set, and the
version number has been bumped accordingly.
Fixes: #740
UserNote: tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
UpgradeNote: A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
(cherry picked from commit 7990f6b589)
Remove duplicate creation of ast_bool_values from
2b7c507d7d12_add_queue_log_option_log_restricted_.py. This was
causing alembic upgrades to fail since the enum was already created
in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018.
Resolves: #797
(cherry picked from commit 59d75aea15)
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.
(cherry picked from commit 41d4db99cf)
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
(cherry picked from commit 7c3e2cfda9)
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
(cherry picked from commit 136bd5c892)
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
(cherry picked from commit 3d40d34271)
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.
(cherry picked from commit 8715a700e2)
This migrates the relevant schema objects from the `('yes', 'no')`
definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')`
one.
Fixes#617
(cherry picked from commit 93e40968c3)
Why do we need a refactor?
The original stir/shaken implementation was started over 3 years ago
when little was understood about practical implementation. The
result was an implementation that wouldn't actually interoperate
with any other stir-shaken implementations.
There were also a number of stir-shaken features and RFC
requirements that were never implemented such as TNAuthList
certificate validation, sending Reason headers in SIP responses
when verification failed but we wished to continue the call, and
the ability to send Media Key(mky) grants in the Identity header
when the call involved DTLS.
Finally, there were some performance concerns around outgoing
calls and selection of the correct certificate and private key.
The configuration was keyed by an arbitrary name which meant that
for every outgoing call, we had to scan the entire list of
configured TNs to find the correct cert to use. With only a few
TNs configured, this wasn't an issue but if you have a thousand,
it could be.
What's changed?
* Configuration objects have been refactored to be clearer about
their uses and to fix issues.
* The "general" object was renamed to "verification" since it
contains parameters specific to the incoming verification
process. It also never handled ca_path and crl_path
correctly.
* A new "attestation" object was added that controls the
outgoing attestation process. It sets default certificates,
keys, etc.
* The "certificate" object was renamed to "tn" and had it's key
change to telephone number since outgoing call attestation
needs to look up certificates by telephone number.
* The "profile" object had more parameters added to it that can
override default parameters specified in the "attestation"
and "verification" objects.
* The "store" object was removed altogther as it was never
implemented.
* We now use libjwt to create outgoing Identity headers and to
parse and validate signatures on incoming Identiy headers. Our
previous custom implementation was much of the source of the
interoperability issues.
* General code cleanup and refactor.
* Moved things to better places.
* Separated some of the complex functions to smaller ones.
* Using context objects rather than passing tons of parameters
in function calls.
* Removed some complexity and unneeded encapsuation from the
config objects.
Resolves: #351Resolves: #46
UserNote: Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
UpgradeNote: The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
(cherry picked from commit e6c7f1aee0)
This adds a dummy migration to 18 and 20 so that our alembic heads are
synchronized across all supported branches.
In this case the migration we are stubbing (24c12d8e9014) is:
775352ee6c
(cherry picked from commit d122225279)
Adding a NoOp alembic script for the voicemail database to maintain
version sync with other branches.
Fixes: #527
(cherry picked from commit f5d622413a)
Instead of searching for the asterisk binary and the modules in the
filesystem, we now get their locations, along with libdir, from
the coredump itself...
For the binary, we can use `gdb -c <coredump> ... "info proc exe"`.
gdb can print this even without having the executable and symbols.
Once we have the binary, we can get the location of the modules with
`gdb ... "print ast_config_AST_MODULE_DIR`
If there was no result then either it's not an asterisk coredump
or there were no symbols loaded. Either way, it's not usable.
For libdir, we now run "strings" on the note0 section of the
coredump (which has the shared library -> memory address xref) and
search for "libasteriskssl|libasteriskpj", then take the dirname.
Since we're now getting everything from the coredump, it has to be
correct as long as we're not crossing namespace boundaries like
running asterisk in a docker container but trying to run
ast_coredumper from the host using a shared file system (which you
shouldn't be doing).
There is still a case for using --asterisk-bin and/or --libdir: If
you've updated asterisk since the coredump was taken, the binary,
libraries and modules won't match the coredump which will render it
useless. If you can restore or rebuild the original files that
match the coredump and place them in a temporary directory, you can
use --asterisk-bin, --libdir, and a new --moddir option to point to
them and they'll be correctly captured in a tarball created
with --tarball-coredumps. If you also use --tarball-config, you can
use a new --etcdir option to point to what normally would be the
/etc/asterisk directory.
Also addressed many "shellcheck" findings.
Resolves: #445
(cherry picked from commit 44f1522907)
`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)
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)
If the safe_asterisk script detects that the /var/lib/asterisk
directory doesn't exist, it now creates it with 755 permissions
instead of 770. safe_asterisk needing to create that directory
should be extremely rare though because it's normally created
by 'make install' which already sets the permissions to 755.
Resolves: #316
(cherry picked from commit c929146c61)
Add quoting around the ps_endpoints 100rel column in the ALTER
statements. Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).
Resolves: #274
(cherry picked from commit cb9223cdb9)
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.
Resolves: #37
(cherry picked from commit fc57dca50d)
In a handful of migrations, the comment header that indicates the
current and previous revisions has drifted from the identifiers
revision and down_revision variables. This updates the comment headers
to match the code.
(cherry picked from commit ea63148b23)
Adds the loop_last option to res_musiconhold,
which allows the last audio file in the directory
to be looped perpetually once reached, rather than
circling back to the beginning again.
Resolves: #122
ASTERISK-30462
UserNote: The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
(cherry picked from commit edf488c76e)
When Asterisk is restarted it does not preserve paused reason for
members of realtime queues. This was fixed for non-realtime queues in
ASTERISK_25732
Resolves: #66
UpgradeNote: Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.
UserNote: Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
(cherry picked from commit df774619fb)
The Asterisk logrotate script contains explicit
references to files with the .log extension,
which are also included when *log is expanded.
This causes issues with newer versions of logrotate.
This fixes this by ensuring that a log file cannot
be referenced multiple times after expansion occurs.
Resolves: #96
ASTERISK-30442
Reported by: EN Barnett
Tested by: EN Barnett
(cherry picked from commit 22c9d52289)
* Remove .gitreview and switch to pulling the main asterisk branch
version from configure.ac instead.
* Replace references to JIRA with GitHub.
* Other minor cleanup found along the way.
Resolves: #39
(cherry picked from commit 5c6d5ea38f)
`rc.archlinux.asterisk`, which explicitly requests bash in its
shebang, uses the following command syntax:
${DAEMON} -rx "core stop now" > /dev/null 2&>1
The intent of which is to execute:
${DAEMON} -rx "core stop now"
While sending both stdout and stderr to `/dev/null`. Unfortunately,
because the `&` is in the wrong place, bash is interpreting the `2` as
just an additional argument to the `$DAEMON` command and not as a file
descriptor and proceeds to use the bashism `&>` to send stderr and
stdout to a file named `1`.
So we clean it up and just use bash's shortcut syntax.
Issue raised and a fix suggested (but not used) by peutch on GitHub¹.
ASTERISK-30449 #close
1. https://github.com/asterisk/asterisk/pull/31
Change-Id: Ie279bf4efb4d95cbf507313483d316e977303d19
(cherry picked from commit 46bdd5e3be)
