In function ast_say_date_with_format_de(), take special
care when the hour is one o'clock. In this case, the
German number "eins" must be inflected to its neutrum form,
"ein". This is achieved by playing "digits/1N" instead of
"digits/1". Fixes both 12- and 24-hour formats.
ASTERISK-30092
Change-Id: Ica9b80125c0b317e378d89c1ea786816e2635510
If a switch is invoked using chan_iax2, deadlock can result
because the PBX core is autoservicing the channel while chan_iax2
also then attempts to service it while waiting for the result
of the switch. This removes servicing of the channel to prevent
any conflicts.
ASTERISK-30064 #close
Change-Id: Ie92f206d32f9a36924af734ddde652b21106af22
If tab completion using ast_module_helper is attempted
during startup, deadlock will ensue because the CLI
will attempt to lock the module list while it is already
locked by the loader. This causes deadlock because when
the loader tries to acquire the CLI lock, they are blocked
on each other.
Waiting for startup to complete is not feasible because
the CLI lock is acquired while waiting, so deadlock will
ensure regardless of whether or not a lock on the module
list is attempted.
To prevent deadlock, we immediately abort if tab completion
is attempted on the module list before Asterisk is fully
booted.
ASTERISK-30039 #close
Change-Id: Idd468906c512bb196631e366a8f597a0e2e9271d
res_calendar will trigger an assertion currently
if the ending time is calculated to be in the past.
Unlike the reminder and start times, however, there
is currently no check to catch non-positive times
and set them to 1. As a result, if we get a negative
value by happenstance, this can cause a crash.
To prevent the assertion from begin triggered, we now
use the same logic as the reminder and start events
to catch this issue before it can cause a problem.
ASTERISK-29981 #close
Change-Id: Idfb3204d195f350d2575fb4bc72a54a597d6e93c
Emits a warning if the user has requested a parking spot that
is out of bounds for the requested parking lot.
ASTERISK-30086
Change-Id: I1080371e4f63e94724455003753014fbd3f95fbf
When a PJSIP channel is set on hold or off hold, all streams were set
on/off hold. This is not the desired behaviour and caused issues
when there were multiple streams in the topology.
Now, only the default audio stream is set on/off hold when a hold is
indicated.
ASTERISK-30051
Change-Id: I04f1110565fd05fea565f5539b534b54549d4f71
The change "Add LOCAL/REMOTE tags in dialog-info+xml" set both "local"
Identity Element URI and Target Element URI to the same value -
the channel Caller Number.
For Identity Element it's ok to set as Caller ID.
But Local Target URI should be set as local URI.
In this case the Local Target URI can be used for Directed Call Pickup
by Polycom ip-phones (parameter useLocalTargetUriforLegacyPickup).
Also XML sanitized Display names.
ASTERISK-24601
Change-Id: If130a2f2f3b2339b14dca0ec0ebeea3a87b34343
Agi commnad exec can now evaluate dialplan functions and
variables if variable AGIEXECFULL is set to yes. this can
be useful when executing Playback or Read from agi.
ASTERISK-30058 #close
Change-Id: I669991f540496e7bddd096fec82b52c083036832
This change exposes the channel driver's unique id (i.e. the Call-ID
for chan_sip/chan_pjsip based channels) to ARI channel resources
as `protocol_id`.
ASTERISK-30027
Reported by: Moritz Fain
Tested by: Moritz Fain
Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87
As part of PJSIP 2.11 a behavior change was done to require
a matching remote hostname on an established transport for
secure transports. Since the Websocket transport is considered
a secure transport this caused the existing connection to not
be found and used.
We now set the remote hostname and the transport can be found.
ASTERISK-30065
Change-Id: Ia1cdef33e1411f927985b4b852c95e163c080e94
This is needed to be able to restore it in REGISTER responses,
otherwise the client won't be able to find the contact it created.
ASTERISK-30042
Change-Id: I0c5823918199acf09246b3b206fbde66773688f6
Adjusts the pjsip show registration(s) commands to show
the amount of seconds remaining until a registration
expires.
ASTERISK-29845 #close
Change-Id: Ic4fea15a1a1056c424416def49d1ca8e776c0483
Adds the CONFBRIDGE_CHANNELS function which can be used
to retrieve a comma-separated list of channels, filtered
by a particular type of participant category. This output
can then be used with functions like UNSHIFT, SHIFT, POP,
etc.
ASTERISK-30036 #close
Change-Id: I1950aff932437476dc1abab6f47fb4ac90520b83
Currently, the operator services mode in DAHDI is broken and unusable.
The actual operator recall functionality works properly; however,
when the operator hangs up (which is the only way that such a call
is allowed to end), both lines are permanently taken out of service
until "dahdi restart" is run. This prevents this feature from being
used.
Operator mode is one of the few factors that can cause the general
analog event handling in sig_analog not to be used. Several years
back, much of the analog handling was moved from chan_dahdi to
sig_analog. However, this was not done fully or consistently at
the time, and when operator mode is active, sig_analog does not
get used. Generally this is correct, but in the case of hangup
it should be using sig_analog regardless of the operator mode;
otherwise, the lines do not properly clear and they become unusable.
This bug is fixed so the operator can now hang up and properly
release the call. It is treated just like any other hangup. The
operator mode functionality continues to work as it did before.
ASTERISK-29993 #close
Change-Id: Ib2e3ddb40d9c71e8801e0b4bb0a12e2b52f51d24
Most issues were in stringfields and had to do with comparing
a pointer to an constant/interned string with NULL. Since the
string was a constant, a pointer to it could never be NULL so
the comparison was always "true". gcc now complains about that.
There were also a few issues where determining if there was
enough space for a memcpy or s(n)printf which were fixed
by defining some of the involved variables as "volatile".
There were also a few other miscellaneous fixes.
ASTERISK-30044
Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570
GCC 12 caught an issue in state_id_by_topic where we were
checking a pointer for NULL instead of the contents of
the pointer for '\0'.
ASTERISK-30044
Change-Id: Ia0b04d4fff45c92acb7f07132a33622fa341148e
When a new unreal (local) channel is created, a second (;2) channel is
created as a counterpart which clones the topology of the first
channel. This creates issues when an outgoing stream is sendonly or
recvonly as the stream state of the inbound channel will be the same
as the stream state of the outbound channel.
Now the stream state is flipped for the streams of the 2nd channel in
ast_unreal_new_channels if the outgoing stream topology is recvonly or
sendonly.
ASTERISK-29655
Reported by: Michael Auracher
ASTERISK-29638
Reported by: Michael Auracher
Change-Id: I0cea29635bb20b7bf7fd0fb95498cd44dab98fbf
Documents the Dial syntax for DAHDI, namely the channel group,
distinctive ring, answer confirmation, and digital call options
that are specified in the resource itself.
ASTERISK-24827 #close
Change-Id: Ib95e78497fb00dc5cbfde1c93a69f034bfd08c30
For lines that have mailboxes configured on them, with
FSK MWI, DAHDI will periodically try to dispatch FSK
to update MWI. However, this is never supposed to be
done when a channel is not idle.
There is currently an edge case where MWI FSK can
extraneously get spooled for the channel if a caller
hook flashes and hangs up, which triggers a recall ring.
After one ring, the on hook time threshold in this if
condition has been satisfied and an MWI update is spooled.
This means that when the phone is picked up again, the
answerer gets an FSK spill before being reconnected to
the party on hold.
To prevent this, we now explicitly check to ensure that
subchannel 0 has no owner. There is no owner when DAHDI
channels are idle, but if the channel is "in use" in some
way (such as in the aforementioned scenario), then there
is an owner, and we shouldn't process MWI at this time.
ASTERISK-28518 #close
Change-Id: Ia3904434fd81688d71742f7e84358b7e1c38e92a
Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio
file.
ASTERISK-29931
Added by Michael Cargile
Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2
Currently, if any custom ring cadences are specified, they are
appended to the array of cadences from wherever we left off
last time. This works properly the first time, but on subsequent
dahdi restarts, it means that the existing cadences are left
alone and (most likely) the same cadences are then re-added
afterwards. In short order, the cadence array gets maxed out
and the user begins seeing warnings that the array is full
and no more cadences may be added.
This buggy behavior persists until Asterisk is completely
restarted; however, if and when dahdi restart is run again,
then the same problem is reintroduced.
This fixes this behavior so that cadence parsing is more
idempotent, that is so running dahdi restart multiple times
starts adding cadences from the beginning, rather than from
wherever the last cadence was added.
As before, it is still not possible to revert to the default
cadences by simply removing all cadences in this manner, nor
is it possible to delete existing cadences. However, this
does make it possible to update existing cadences, which
was not possible before, and also ensures that the cadences
remain unchanged if the config remains unchanged.
ASTERISK-29990 #close
Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d
Currently, if attempting to place a call to a peer that only allows
RSA authentication, if we fail to provide an outkey when placing
the call, Asterisk will crash.
This exposes the broader issue that IAX2 is prone to causing a crash
if encryption or decryption is attempted but we never initialized
the encryption and decryption keys. In other words, if the logic
to use encryption in chan_iax2 is not perfectly aligned with the
decision to build keys in the first place, then a crash is not
only possible but probable. This was demonstrated by ASTERISK_29264,
for instance.
This permanently prevents such events from causing a crash by explicitly
checking that keys are initialized properly before setting the flags
to use encryption for the call. Instead of crashing, the call will
now abort.
ASTERISK-30007 #close
Change-Id: If925c3d86099ceac7f621804f2532baac5050c9a
A bug in menuselect can cause modules that are disabled
by default to be recompiled every time a recompilation
occurs. This occurs for module categories that are NOT
positive output, as for these categories, the modules
contained in the makeopts file indicate modules which
should NOT be selected. The existing procedure of iterating
through these modules to mark modules as present is thus
insufficient. This has led to modules with a default_enabled
tag of "no" to get deleted and recompiled every time, even
when they haven't changed.
To fix this, we now modify the mark as present behavior
for module categories that are not positive output. For
these, we start by iterating through the module tree
and marking all modules as present, then go back and
mark anything contained in the makeopts file as not
present. This ensures that makeopt selections are actually
used properly, regardless of whether a module category
uses positive output or not.
ASTERISK-29728 #close
Change-Id: Idf2974c4ed8d0ba3738a92f08a6082b234277b95
The admin_exec function in app_meetme is used by the SLA
applications for internal bridging. However, in these cases,
chan is NULL. Currently, this function will set some status
variables that are intended for a channel, but since channel
is NULL, this is erroneously creating meaningless global
variables, which shouldn't be happening. This sets these
variables only if chan is not NULL.
ASTERISK-30002 #close
Change-Id: I817df6c26f5bda131678e56791b0b61ba64fc6f7
Some command line options to Asterisk only apply when Asterisk
is started and cannot be used with remote console mode. If a
user tries to use any of these, they are currently simply
silently ignored.
This prints out a warning if incompatible options are used,
informing users that an option used cannot be used with remote
console mode. Additionally, some clarifications are added to
the help text and man page.
ASTERISK-22246
ASTERISK-26582
Change-Id: I980a5380ef2c19e8ea348596396d5382893c4337
Adds the DB_KEYCOUNT function, which can be used to retrieve
the number of keys at a given prefix in AstDB.
ASTERISK-29968 #close
Change-Id: Ib2393b77b7e962dbaae6192f8576bc3f6ba92d09
According to chan_dahdi.conf, up to 64 groups (numbered
0 through 63) can be used when dialing DAHDI channels.
However, currently dialing round robin with a group number
greater than 31 fails because the array for the round robin
structure is only size 32, instead of 64 as it should be.
This fixes that so the round robin array size is consistent
with the actual groups capacity.
ASTERISK-29994
Change-Id: I4caa08d7025f78ac75a0539f71aaf3eb3e85b3b7
If Asterisk receives a SIP REFER with Session-Timers UAC
maintain Session-Timers when sending UPDATE"
ASTERISK-29843
Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c
This adds the EVAL_EXTEN function, which may be used to retrieve
the variable-substituted data at any extension.
ASTERISK-29486
Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432
Currently, if a user uses an application like ControlPlayback
to try to rewind a file past the beginning, this can throw
warnings when the file format (e.g. PCM) tries to seek to
a negative offset.
Instead of letting file formats try (and fail) to seek a
negative offset, we instead now catch this in the rewind
function to ensure that we never seek an offset less than 0.
This prevents legitimate user actions from triggering warnings
from any particular file formats.
ASTERISK-29943 #close
Change-Id: Ia53f2623f57898f4b8e5c894b968b01e95426967
PJSIP currently is capable of receiving flash events
and converting them to FLASH control frames, but it
currently lacks support for doing the reverse: taking
a FLASH control frame and converting it into a flash
event in the SIP domain.
This adds the ability for PJSIP to process flash control
frames by converting them into the appropriate SIP INFO
message, which can then be sent to the peer. This allows,
for example, flash events to be sent between Asterisk
systems using PJSIP.
ASTERISK-29941 #close
Change-Id: I1590221a4d238597f79672fa5825dd4a920c94dd
Adds the dialplan eval function commands to evaluate a dialplan
function from the CLI. The return value and function result are
printed out and can be used for testing or debugging.
ASTERISK-29820 #close
Change-Id: I833e97ea54c49336aca145330a2adeebfad05209
Adds version information for applications, functions,
and manager events/actions.
This is not completely exhaustive by any means but
covers most new things added that have release
versioning information in the issue tracker.
ASTERISK-29940 #close
Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131
Removes a couple sample config files for modules
which have since been removed from Asterisk.
ASTERISK-30008 #close
Change-Id: I6be89cafc6c575d98a5315e4912b61a833aacf52
added new global config option "allow_sending_180_after_183"
that if enabled will preserve 180 after a 183
ASTERISK-29842
Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18
if Asterisk need to send an UPDATE before answer
on a channel that uses Record-Route:
it will not include a Route header
ASTERISK-29955
Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef
On a write error to an AMI session a flag was set to
indicate that the write error had occurred, with the
expected result being that the session be terminated.
This was not actually happening and instead writing
would continue to be attempted.
This change adds a check for the write error and causes
the session to actually terminate.
ASTERISK-29948
Change-Id: Icaf5d413d4c0d5dc78292a17287fecc8720a31a5
Patch provided inline by Yury Kirsanov on the linked issue and
approved by Josh Colp.
ASTERISK-29253 #close
Change-Id: I5b9ccc67ebf06e875ed061d9e7fc21f47b0a4e1f
Add framework to connect to, and read and write protocol based
messages from and to an external application using an Asterisk
External Application Protocol (AEAP). This has been divided into
several abstractions:
1. transport - base communication layer (currently websocket only)
2. message - AEAP description and data (currently JSON only)
3. transaction - links/binds requests and responses
4. aeap - transport, message, and transaction handler/manager
This patch also adds an AEAP implementation for speech to text.
Existing speech API callbacks for speech to text have been completed
making it possible for Asterisk to connect to a configured external
translator service and provide audio for STT. Results can also be
received from the external translator, and made available as speech
results in Asterisk.
Unit tests have also been created that test the AEAP framework, and
also the speech to text implementation.
ASTERISK-29726 #close
Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2
When executing dial, the topology of the incoming channel is cloned and
used for the outgoing channel. This creates issues when an incoming
stream is sendonly or recvonly as the stream state of the outgoing
channel will be the same as the stream state of the incoming channel.
Now the stream state is flipped for the outgoing stream in
dial_exec_full if the incoming stream topology is recvonly or sendonly.
ASTERISK-29655
Reported by: Michael Auracher
ASTERISK-29638
Reported by: Michael Auracher
Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1
There was an issue with the conditional where STIR/SHAKEN would be
enabled even when not configured. It has been changed to ensure that if
a profile does not exist and stir_shaken is not set in pjsip.conf, then
the conditional will return from the function without performing
STIR/SHAKEN operations.
ASTERISK-30024
Change-Id: I41286a3d35b033ccbfbe4129427a62cb793a86e6
The async_operations setting on a transport configures how
many simultaneous incoming packets the transport can handle
when multiple threads are polling and waiting on the transport.
As we only use a single thread this was needlessly creating
incoming packets when set to a non-default value, wasting memory.
ASTERISK-30006
Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f
Chrome has added more attributes, causing the limit to be
exceeded. This raises it up some more.
ASTERISK-30015
Change-Id: I964957c005c4e6f7871b15ea1ccd9b4659c7ef32
Adds a new configuration option, stir_shaken_profile, in pjsip.conf that
can be specified on a per endpoint basis. This option will reference a
stir_shaken_profile that can be configured in stir_shaken.conf. The type
of this option must be 'profile'. The stir_shaken option can be
specified on this object with the same values as before (attest, verify,
on), but it cannot be off since having the profile itself implies wanting
STIR/SHAKEN support. You can also specify an ACL from acl.conf (along
with permit and deny lines in the object itself) that will be used to
limit what interfaces Asterisk will attempt to retrieve information from
when reading the Identity header.
ASTERISK-29476
Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406
Put checks in place to limit how much we will actually download, as well
as a check for the data we receive at the start to ensure it begins with
what we would expect a certificate to begin with.
ASTERISK-29872
Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46
Some databases depending on their configuration using backslashes
for escaping. When combined with the use of ' this can result in
a broken func_odbc query.
This change adds a SQL_ESC_BACKSLASHES dialplan function which can
be used to escape the backslashes.
This is done as a dialplan function instead of being always done
as some databases do not require this, and always doing it would
result in incorrect data being put into the database.
ASTERISK-29838
Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d
