Added needed UTF-8 checks before constructing json objects in various
files for strings obtained outside the system. In this case string values
from a channel driver's peer and not from the user setting channel
variables.
* aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
object construction.
ASTERISK-26466
Reported by: Richard Mudgett
Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096
* Updated unit test as ast_json_name_number() is now NULL tolerant.
ASTERISK-26466 #close
Reported by: Richard Mudgett
Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6
Since the json library does not make the check function public we
recreate/copy the function in our interface module.
ASTERISK-26466
Reported by: Richard Mudgett
Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99
* In s_to_json() removed unnecessary ast_json_ref() to ast_json_null()
when creating the type json object. The ref is a noop.
Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a
Remote asterisk consoles should only display verbose log messages
created by the daemon. The first patch for ASTERISK-26410 caused
a couple verbose messages to be printed when the rasterisk process
ended.
ASTERISK-26410
Change-Id: Ie2a1bb3753ad2724c0349ec1a336f52f7117b52a
The main frame read and write handlers in main/channel.c don't use the
optimum placement in the processing flow for calling audiohooks
callbacks, as far as codec translation is concerned. This change places
the audiohooks callback code:
* After the channel read translation if the frame is not linear before
the translation, thereby increasing the chance that the frame is linear
as required by audiohooks
* Before the channel write translation if the frame is linear at this
point
This prevents the audiohooks code from instantiating additional
translation paths to/from linear where a linear frame format is already
available, saving valuable CPU cycles
ASTERISK-26419
Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f
This change introduced some fax test failures
that have not yet been addressed. So this is
not forgotten I'm submitting a change which
reverts it.
This reverts:
d56fc3b36b.
ASTERISK-25629
Change-Id: Ibc2f23c38643f5a2c89cf8915ae2d805b81bc3d5
Asterisk only supports mono audio at the moment.
This patch adds interleaved two-channel audio to Asterisk's channels.
ASTERISK-26292
Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a
* Compile __ast_assert_failed unconditionally.
* Use __ast_assert_failed to log messages from log_bad_ao2
* Remove calls to ast_assert(0) that happen after log_bad_ao2 was run.
Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751
This allows asterisk to compiled with LOW_MEMORY to load modules built
without LOW_MEMORY.
ASTERISK-26398 #close
Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d
Verbose messages should be printed to the console if the sublevel is
less than option_verbose. This fix ensures the welcome message with
copyright and license are printed at daemon and interactive rasterisk
startup.
ASTERISK-26410 #close
Change-Id: Ia44235e30ec328aba92ea2c8a837b094e65c9a03
Users upgrading from asterisk 13.5 to a later version and who use
realtime with peers that have mailboxes were experiencing runaway
situations that manifested as a continuous stream of taskprocessor
congestion errors, memory leaks and an unresponsive chan_sip.
A related issue was that setting rtcachefriends=no NEVER worked in
asterisk 13 (since the move to stasis). In 13.5 and earlier, when a
peer tried to register, all of the stasis threads would block and
chan_sip would again become unresponsive. After 13.5, the runaway
would happen.
There were a number of causes...
* mwi_event_cb was (indirectly) calling build_peer even though calls to
mwi_event_cb are often caused by build_peer.
* In an effort to prevent chan_sip from being unloaded while messages
were still in flight, destroy_mailboxes was calling
stasis_unsubscribe_and_join but in some cases waited forever for the
final message.
* add_peer_mailboxes wasn't properly marking the existing mailboxes
on a peer as "keep" so build_peer would always delete them all.
* add_peer_mwi_subs was unsubscribing existing mailbox subscriptions
then just creating them again.
All of this was causing a flood of subscribes and unsubscribes on
multiple threads all for the same peer and mailbox.
Fixes...
* add_peer_mailboxes now marks mailboxes correctly and build_peer only
deletes the ones that really are no longer needed by the peer.
* add_peer_mwi_subs now only adds subscriptions marked as "new" instead
of unsubscribing and resubscribing everything. It also adds the peer
object's address to the mailbox instead of its name to the subscription
userdata so mwi_event_cb doesn't have to call build_peer.
With these changes, with rtcachefriends=yes (the most common setting),
there are no leaks, locks, loops or crashes at shutdown.
rtcachefriends=no still causes leaks but at least it doesn't lock, loop
or crash. Since making rtcachefriends=no work wasnt in scope for this
issue, further work will have to be deferred to a separate patch.
Side fixes...
* The ast_lock_track structure had a member named "thread" which gdb
doesn't like since it conflicts with it's "thread" command. That
member was renamed to "thread_id".
ASTERISK-25468 #close
Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0
When retrieving presence state information there is no
guarantee that the subtype and message passed in are
set to NULL. This change ensures they are.
ASTERISK-26397 #close
Change-Id: If38cd730e409e9a9b6eb9adef6591d15a9e61f86
Routines responsible for managing ast_callid's are overly complicated.
This is left-over code from when ast_callid was an AO2 object. Now that
it is an integer the code can be reduced.
ast_callid handler code no longer prints it's own error message upon failure
to allocate threadstorage as ast_calloc would have already printed a
message. Debug messages that were printed when TEST_FRAMEWORK was
enabled have been also been removed.
Change-Id: I65a768a78dc6cf3cfa071e97f33ce3dce280258e
Previous versions of Asterisk did not require verbose to be specified in
logger.conf for the console channel, if it was requested by command line
or asterisk.conf it just worked. This change causes Asterisk to always
enable verbose in the console channel level mask. Verbose is displayed
on consoles if requested by command line, option_verbose or 'core set
verbose'.
This also delays initialization of the logger until after threadstorage
is initialized. Initializing too early can cause messages to be printed
multiple times to the console (stdout).
ASTERISK-26391 #close
Change-Id: I52187d67c2fcb3efd5561bf04b3e5e23e5ee8a04
When logger.conf is missing or invalid we should be printing notices,
warnings and errors to the console. The logmask was incorrectly
calculated.
Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3
Without this change, a 'core restart' would kill the astcanary forever
if you're not running as root. Both with and without this patch, the
scheduling priority was still SCHED_RR after restart.
Additionally, the astcanary is now spawned if you start with high
priority and Asterisk doesn't get a chance to lower it. For example
through: `chrt -r 10 sudo -u asterisk asterisk -c`
Also reap killed astcanary processes on core restart.
ASTERISK-26352 #close
Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55
Previously only the canary checking thread itself had its priority set
to SCHED_OTHER. Now all threads are traversed and adjusted.
ASTERISK-19867 #close
Reported by: Xavier Hienne
Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39
When receiving an SDP offer with multiple payloads for
the same format we would generate an answer with the first
payload, but during the payload crossover operation
(to set the payloads for receiving) we would remove all
payloads but the last. This would result in incoming
traffic being matched against the wrong format and outgoing
traffic being sent using the wrong payload.
This change makes it so that once a format has a payload
number put into the mapping all subsequent ones are ignored.
This ensures there is only ever one payload in the mapping
and that it is the payload placed into the answer SDP.
ASTERISK-26365 #close
Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60
sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).
To use this, use a systemd unit with 'Type=notify' for Asterisk.
This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.
Also adds support for libsystemd detection in the configure script.
Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811
If sysinfo() is available, but not sysctl() or swapctl() the
printing code for swap buffer sizes is incorrectly omitted.
The above condition happens with musl c-library.
Fix #if rule to consider defined(HAVE_SYSINFO). And also
remove the redundant || defined(HAVE_SYSCTL) which was
incorrectly there to start with. Now swap information is
displayed only if an actual libc function to get it is
available.
This also fixes warnings previously seen with musl libc:
[CC] asterisk.c -> asterisk.o
asterisk.c: In function 'handle_show_sysinfo':
asterisk.c:773:6: warning: variable 'totalswap' set but not used
[-Wunused-but-set-variable]
int totalswap = 0;
^~~~~~~~~
asterisk.c:770:11: warning: variable 'freeswap' set but not used
[-Wunused-but-set-variable]
uint64_t freeswap = 0;
^~~~~~~~
Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca
Currently when receiving video over RTP we store only
a calculated samples on the frame. When starting the video
it can take some time for this calculation to actually yield
a value as it requires constant changing timestamps. As well
if a video frame passes over multiple RTP packets this calculation
will fail as the timestamp is the same as the previous RTP
packet and the number of samples calculated will be 0.
This change preserves the timestamp on the frame and allows
it to pass through the core. When sending the video this timestamp
is used instead of a new one being calculated.
ASTERISK-26367 #close
Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd
Create an alternative to ast_sorcery_generic_alloc which uses astobj2
shared locking. Use this new method for the 'struct ast_sip_aor' allocator.
Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f
This allows standard ao2 functions to be used to release references to
an ast_named_lock. This change can cause less frequent locking of the
global named_locks container. The container is no longer locked when a
named_lock reference is being release except when this causes the
named_lock to be destroyed.
Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6
Previously, if context A was including context B and context B was including
context A, i.e. if there was a circular dependency between contexts, then
calling manager_show_dialplan_helper could lead to an infinite recursion,
resulting in a crash.
This commit applies the same solution as the one implemented in the
show_dialplan_helper function. The manager_show_dialplan_helper and
show_dialplan_helper functions contain lots of code in common, but the former
was missing the "infinite recursion avoidance" code.
ASTERISK-26226 #close
Change-Id: I1aea85133c21787226f4f8442253a93000aa0897
zuul
Richard Mudgett
Corey Farrell
Michael Walton
Joshua Colp
frahaase
George Joseph
Walter Doekes
Tzafrir Cohen
Timo Teräs
Etienne Lessard