When started with a verbose level of 3, asterisk can emit over 1500
verbose message that serve no real purpose other than to fill up
logs. When asterisk shuts down, it emits another 1100 that are of
even less use. Since the testsuite runs asterisk with a verbose
level of 3, and asterisk starts and stops for every one of the 700+
tests, the number of log messages is staggering. Besides taking up
resources, it also makes it hard to debug failing tests.
This commit changes the log level for those verbose messages to 5
instead of 3 which reduces the number of log messages to only a
handful. Of course, NOTICE, WARNING and ERROR message are
unaffected.
There's also one other minor change...
ast_context_remove_extension_callerid2() logs a DEBUG message
instead of an ERROR if the extension you're deleting doesn't exist.
The pjsip_config_wizard calls that function to clean up the config
and has been triggering that annoying error message for years.
Resolves: #582
The last time configure was run, it was run on a system that
did not enable -std=gnu11 by default, which meant that the
restrict qualifier would not be recognized on certain platforms.
This regenerates the configure files from running bootstrap.sh,
so that these should be recognized on all supported platforms.
Resolves: #586
Fixes: #406
UserNote: Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
Adds 'p' option to SpeechBackground() application.
With this option, when the app timeout is reached,
whatever the backend speech engine collected will
be returned as if it were the final, full result.
(This works for engines that make partial results.)
Resolves: #572
UserNote: The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
This introduces a setting for outbound registrations to override the
global User-Agent header setting.
Resolves: #515
UserNote: PJSIP outbound registrations now support a per-registration
User-Agent header
Given the scenario of passing an empty string to the
ast_strsep functions the functions would return NULL
instead of an empty string. This is counter to how
strsep itself works.
This change alters the behavior of the functions to
match that of strsep.
Fixes: #565
Adds the 'D' option to app chanspy that causes the input and output
frames of the spied channel to be interleaved in the spy output frame.
This allows the input and output of the spied channel to be decoded
separately by the receiver.
If the 'o' option is also set, the 'D' option is ignored as the
audio being spied is inherently one direction.
Fixes: #569
UserNote: The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
Commit fa3922a4d2 fixed
a branching issue but "overshoots" when calculating
the next priority. This fixes that; accompanying
test suite tests have also been extended.
Resolves: #560
Resolves a regression identified by @justinludwig involving the
rendering of IPv6 addresses in outgoing SDP.
Also updates `media_address` on PJSIP endpoints so that if we are able
to parse the configured value as an IP we store it in a format that we
can directly use later. Based on my reading of the code it appeared
that one could configure `media_address` as:
```
[foo]
type = endpoint
...
media_address = [2001:db8::]
```
And that value would be blindly copied into the outgoing SDP without
regard to its format.
Fixes#541
Currently, a reload will always occur if the
Reload header is provided for the UpdateConfig
action. However, we should not be doing a reload
if the header value has a falsy value, per the
documentation, so this makes the reload behavior
consistent with the existing documentation.
Resolves: #551
The numeric bridge profile options `internal_sample_rate` and
`maximum_sample_rate` are documented to accept the special values
`auto` and `none`, respectively. While these values currently work,
they also emit warnings when used which could be confusing for users.
In passing, also ensure that we only accept the documented range of
sample rate values between 8000 and 192000.
Fixes#546
When app_macro was deprecated, the macrocontext column was removed from
the INSERT statement but the binds were not renumbered. This broke the
insert.
This change removes the macrocontext column via alembic and re-numbers
the existing columns in the INSERT.
Fixes: #527
UserNote: The fix requires removing the macrocontext column from the
voicemail_messages table in the voicemail database via alembic upgrade.
UpgradeNote: The fix requires that the voicemail database be upgraded via
alembic. Upgrading to the latest voicemail database via alembic will
remove the macrocontext column from the voicemail_messages table.
This adds a CLI command to manually toggle the MWI status
of a channel, useful for troubleshooting or resetting
MWI devices, similar to the capabilities offered with
SIP messaging to manually control MWI status.
UserNote: The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Resolves: #440
Commit 008731b0a4
caused a regression by resulting in logger.xml
being compiled and linked into the asterisk
binary in lieu of logger.c on certain platforms
if Asterisk was compiled in dev mode.
To fix this, we ensure the file has a unique
name without the extension. Most existing .xml
files have been named differently from any
.c files in the same directory or did not
pose this issue.
channels/pjsip/dialplan_functions.xml does not
pose this issue but is also being renamed
to adhere to this policy.
Resolves: #539
This adds a simple CLI command that can be used for
analyzing all frames currently queued to a channel.
A couple log messages are also adjusted to be more
useful in tracing bridging problems.
Resolves: #533
This reverts commit 315eb551db.
Over the past year, we've had several reports of "topology storms"
occurring where 2 external facing channels connected by one or more
local channels and bridges will get themselves in a state where
they continually send each other topology change requests. This
usually manifests itself in no-audio calls and a flood of
"Exceptionally long queue length" messages. It appears that this
commit is the cause so we're reverting it for now until we can
determine a more appropriate solution.
Resolves: #530
This fixes faulty branching logic for the
EndIf application. Instead of computing
the next priority, which should be done
for false conditionals or ExitIf, we should
simply advance to the next priority.
Resolves: #341
Commit 424be34563 introduced
a regression by calling ast_free on memory allocated by
realpath. This causes Asterisk to abort when executing this
function. Since the memory is allocated by glibc, it should
be freed using ast_std_free.
Resolves: #513
* Since ICE candidates are used for the check and pjproject is
required to use ICE, res_rtp_asterisk was failing to compile
when pjproject wasn't available. The check is now wrapped
with an #ifdef HAVE_PJPROJECT.
* The rtp->ice_active_remote_candidates container was being
used to check the address on incoming packets but that
container doesn't contain peer reflexive candidates discovered
during negotiation. This was causing the check to fail
where it shouldn't. We now check against pjproject's
real_ice->rcand array which will contain those candidates.
* Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
we weren't zeroing out sin->sin_zero before returning. This
was causing ast_sockaddr_cmp() to always return false when
one of the inputs was converted from a pj_sockaddr, even
if both inputs had the same address and port.
Resolves: #500Resolves: #503Resolves: #505
When updating an existing header the 'update' code incorrectly
just copied the new value into the existing buffer. If the
new value exceeded the available buffer size memory outside
of the buffer would be written into, potentially causing
a crash.
This change makes it so that the 'update' now duplicates
the new header value instead of copying it into the existing
buffer.
Add patch to split the log level for invalid packets received on the
signaling port. The warning regarding the packet will move to level 2
so that it can still be displayed, while the raw packet will be at level
4.
When ICE is in use, we can prevent a possible DOS attack by allowing
DTLS protocol messages (client hello, etc) only from sources that
are in the active remote candidates list.
Resolves: GHSA-hxj9-xwr8-w8pq
When using AMI GetConfig, it was possible to access files outside of the
Asterisk configuration directory by using filenames with ".." and "./"
even while live_dangerously was not enabled. This change resolves the
full path and ensures we are still in the configuration directory before
attempting to access the file.
Improve the "manager show connected" CLI command
to clarify that the last two columns are permissions
related, not counts, and use sufficient widths
to consistently display these values.
ASTERISK-30143 #close
Resolves: #482
Although `make_xml_documentation`'s `print_dependencies` command was
corrected by the previous fix (#461) for #142, the `create_xml` was
not properly handling `LOCAL_MOD_SUBDIRS` XML documentation.
This fixes a number of broken links throughout the
tree, mostly caused by wiki.asterisk.org being replaced
with docs.asterisk.org, which should eliminate the
need for sporadic fixes as in f28047db36.
Resolves: #430
Resolves: #462
UserNote: The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
`pbx_config` subscribes to manager events to capture the `FullyBooted`
event but fails to unsubscribe if the module is loaded after that
event fires. If the module is unloaded, a crash occurs the next time a
manager event is raised.
We now unsubscribe when the module is unloaded if we haven't already
unsubscribed.
Fixes#470
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
The `get_documentation` awk script will only extract the first
DOCUMENTATION block that it finds in a given file. This is by design
(9bc2127) to prevent AMI event documentation from being pulled in to
the core.xml documentation file.
Because of this, the `LOG_GROUP` documentation added in 89709e2 was
not being properly extracted and was missing fom the resulting XML
documentation file. This commit moves the `LOG_GROUP` documentation to
a separate `logger.xml` file.
There are valid scenarios where res_odbc's connection pool might have some dead
or stuck connections while others are healthy (imagine network
elements/firewalls/routers silently timing out connections to a single DB and a
single IP address, or a heterogeneous connection pool connected to potentially
multiple IPs/instances of a replicated DB using a DNS front end for load
balancing and one replica fails).
In order to time out those unhealthy connections without blocking access to
other parts of Asterisk that may attempt access to the connection pool, it would
be beneficial to not lock/block access around the entire pool in
_ast_odbc_request_obj2 while doing potentially blocking operations on connection
pool objects such as the connection_dead() test, odbc_obj_connect(), or by
dereferencing a struct odbc_obj for the last time and triggering a
odbc_obj_disconnect().
This would facilitate much quicker and concurrent timeout of dead connections
via the connection_dead() test, which could block potentially for a long period
of time depending on odbc.ini or other odbc connector specific timeout settings.
This also would make rapid failover (in the clustered DB scenario) much quicker.
This patch changes the locking in _ast_odbc_request_obj2() to not lock around
odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to
lock around truly shared, non-immutable state like the connection_cnt member and
the connections list on struct odbc_class.
Fixes: #465
If the script referenced by `#exec` does not exist, writes anything to
stderr, or exits abnormally or with a non-zero exit status, we log
that to Asterisk's error logging channel.
Additionally, write out a warning if the script produces no output.
Fixes#259
Resequencing is a process that occurs when we open a voicemail folder
and discover that there are gaps between messages (e.g. `msg0000.txt`
is missing but `msg0001.txt` exists). Resequencing involves shifting
the existing messages down so we end up with a sequential list of
messages.
Currently, this process stops after reaching a threshold based on the
message limit (`maxmsg`) configured on the current folder. However, if
`maxmsg` is lowered when a voicemail folder contains more than
`maxmsg + 10` messages, resequencing will not run completely leaving
the mailbox in an inconsistent state.
We now resequence up to the maximum number of messages permitted by
`app_voicemail` (currently hard-coded at 9999 messages).
Fixes#86
When mwimonitor=yes is enabled for an FXO port,
the do_monitor thread will launch mwi_thread if it thinks
there could be MWI on an FXO channel, due to the noise
threshold being satisfied. This, in turns, calls
analog_ss_thread_start in sig_analog. However, unlike
all other instances where __analog_ss_thread is called
in sig_analog, this call path does not properly set
pvt->ss_astchan to the Asterisk channel, which means
that the Asterisk channel is NULL when __analog_ss_thread
starts executing. As a result, the thread exits and the
channel is never properly cleaned up by calling ast_hangup.
This caused issues with do_monitor on incoming calls,
as it would think the channel was still owned even while
receiving events, leading to an infinite barrage of
warning messages; additionally, the channel would persist
improperly.
To fix this, the assignment is added to the call path
where it is missing (which is only used for mwi_thread).
A warning message is also added since previously there
was no indication that __analog_ss_thread was exiting
abnormally. This resolves both the channel leak and the
condition that led to the warning messages.
Resolves: #458
George Joseph
Naveen Albert
Ben Ford
cmaj
Flole998
Joshua C. Colp
Mike Bradeen
Sean Bright
Brad Smith
PeterHolik
Maximilian Fridrich
Gitea
Matthew Fredrickson