Naveen Albert [Sun, 24 Oct 2021 18:38:13 +0000 (18:38 +0000)]
res_pjsip_callerid: Fix OLI parsing
Fix parsing of ANI2/OLI information, since it was previously
parsing the user, when it should have been parsing other_param.
Also improves the parsing by using pjproject native functions
rather than trying to parse the parameters ourselves like
chan_sip did. A previous attempt at this caused a crash, but
this works correctly now.
Currently, when the t option is specified with no arguments,
the # character is still treated as a terminator, even though
no character should be treated as a terminator.
This is because a previous regression fix was modified to
remove the use of NULL as a default altogether. However,
NULL and an empty string actually refer to different
arrangements and should be treated differently. NULL is the
default terminator (#), while an empty string removes the
terminator altogether. This is the behavior being used by
the rest of the core.
Additionally, since S_OR catches empty strings as well as
NULL (not intended), this is changed to a ternary operator
instead, which fixes the behavior.
Naveen Albert [Sun, 24 Oct 2021 11:55:50 +0000 (11:55 +0000)]
sig_analog: Fix truncated buffer copy
Fixes compiler warning caused by a truncated copy of the ANI2 into a
buffer of size 10. This could prevent the null terminator from being
copied if the copy value exceeds the size of the buffer. This increases
the buffer size to 101 to ensure there is no way for truncation to occur.
Sean Bright [Mon, 8 Nov 2021 15:01:09 +0000 (10:01 -0500)]
pbx.c: Don't remove dashes from hints on reload.
When reloading dialplan, hints created dynamically would lose any dash
characters. Now we ignore those dashes if we are dealing with a hint
during a reload.
Naveen Albert [Sun, 24 Oct 2021 12:31:37 +0000 (12:31 +0000)]
app_voicemail: Fix phantom voicemail bug on rerecord
If users are able to press # for options while leaving
a message and then press 3 to rerecord the message, if
the caller hangs up during the rerecord prompt but before
Asterisk starts recording a message, then an "empty"
voicemail gets processed whereby an email gets sent out
notifying the user of a 0:00 duration message. The file
doesn't actually exist, so playback will fail since there
was no message to begin with.
This adds a check after the streaming of the rerecord
announcement to see if the caller has hung up. If so,
we bail out early so that we can clean up properly.
Naveen Albert [Tue, 26 Oct 2021 00:47:02 +0000 (00:47 +0000)]
chan_iax2: Allow both secret and outkey at dial time
Historically, the dial syntax for IAX2 has held that
an outkey (used only for RSA authenticated calls)
and a secret (used only for plain text and MD5 authenticated
calls, historically) were mutually exclusive, and thus
the same position in the dial string was used for both
values.
Now that encryption is possible with RSA authentication,
this poses a limitation, since encryption requires a
secret and RSA authentication requires an outkey. Thus,
the dial syntax is extended so that both a secret and
an outkey can be specified.
The new extended syntax is backwards compatible with the
old syntax. However, a secret can now be specified after
the outkey, or the outkey can be specified after the secret.
This makes it possible to spawn an encrypted RSA authenticated
call without a corresponding peer being predefined in iax.conf.
George Joseph [Thu, 9 Sep 2021 14:39:27 +0000 (08:39 -0600)]
ast_coredumper: Refactor to better find things
The search for a running asterisk when --running is used
has been greatly simplified and in the event it doesn't
work, you can now specify a pid to use on the command
line with --pid.
The search for asterisk modules when --tarball-coredumps
is used has been enhanced to have a better chance of finding
them and in the event it doesn't work, you can now specify
--libdir on the command line to indicate the library directory
where they were installed.
The DATEFORMAT variable was renamed to DATEOPTS and is now
passed to the 'date' utility rather than running DATEFORMAT
as a command.
The coredump and output files are now renamed with DATEOPTS.
This can be disabled by specifying --no-rename.
Several confusing and conflicting options were removed:
--append-coredumps
--conffile
--no-default-search
--tarball-uniqueid
The script was re-structured to make it easier for follow.
Ben Ford [Tue, 21 Sep 2021 17:09:10 +0000 (12:09 -0500)]
STIR/SHAKEN: Option split and response codes.
The stir_shaken configuration option now has 4 different choices to pick
from: off, attest, verify, and on. Off and on behave the same way they
do now. Attest will only perform attestation on the endpoint, and verify
will only perform verification on the endpoint.
Certain responses are required to be sent based on certain conditions
for STIR/SHAKEN. For example, if we get a Date header that is outside of
the time range that is considered valid, a 403 Stale Date response
should be sent. This and several other responses have been added.
Kevin Harwell [Thu, 21 Oct 2021 17:49:44 +0000 (12:49 -0500)]
res_speech: Add a type conversion, and new engine unregister methods
Add a new function that converts a speech results type to a string.
Also add another function to unregister an engine, but returns a
pointer to the unregistered engine object instead of a success/fail
integer.
Mike Bradeen [Thu, 7 Oct 2021 18:07:24 +0000 (12:07 -0600)]
various: Fix GCC 11 compilation issues.
test_voicemail_api: Use empty char* for empty_msg_ids.
chan_skinny: Fix size of calledParty to be maximum extension.
menuselect: Change Makefile to stop deprecated warnings. Added comments
test_linkedlist: 'bogus' variable was manually allocated from a macro
and the test fails if this happens but the compiler couldn't 'see' this
and returns a warning. memset to all 0's after allocation.
chan_ooh323: Fixed various indentation issues that triggered misleading
indentation warnings.
apps/app_playback.c: Add 'mix' option to app_playback
I am adding a mix option that will play by filename and say.conf unlike
say option that will only play with say.conf. It
will look on the format of the name, if it is like say it play with
say.conf if not it will play the file name.
George Joseph [Tue, 19 Oct 2021 16:35:26 +0000 (10:35 -0600)]
BuildSystem: Check for alternate openssl packages
OpenSSL is one of those packages that often have alternatives
with later versions. For instance, CentOS/EL 7 has an
openssl package at version 1.0.2 but there's an openssl11
package from the epel repository that has 1.1.1. This gets
installed to /usr/include/openssl11 and /usr/lib64/openssl11.
Unfortunately, the existing --with-ssl and --with-crypto
./configure options expect to point to a source tree and
don't work in this situation. Also unfortunately, the
checks in ./configure don't use pkg-config.
In order to make this work with the existing situation, you'd
have to run...
./configure --with-ssl=/usr/lib64/openssl11 \
--with-crypto=/usr/lib64/openssl11 \
CFLAGS=-I/usr/include/openssl11
BUT... those options don't get passed down to bundled pjproject
so when you run make, you have to include the CFLAGS again
which is a big pain.
Oh... To make matters worse, although you can specify
PJPROJECT_CONFIGURE_OPTS on the ./configure command line,
they don't get saved so if you do a make clean, which will
force a re-configure of bundled pjproject, those options
don't get used.
So...
* In configure.ac... Since pkg-config is installed by install_prereq
anyway, we now use it to check for the system openssl >= 1.1.0.
If that works, great. If not, we check for the openssl11
package. If that works, great. If not, we fall back to just
checking for any openssl. If pkg-config isn't installed for some
reason, or --with-ssl=<dir> or --with-crypto=<dir> were specified
on the ./configure command line, we fall back to the existing
logic that uses AST_EXT_LIB_CHECK().
* The whole OpenSSL check process has been moved up before
THIRD_PARTY_CONFIGURE(), which does the initial pjproject
bundled configure, is run. This way the results of the above
checks, which may result in new include or library directories,
is included.
* Although not strictly needed for openssl, We now save the value of
PJPROJECT_CONFIGURE_OPTS in the makeopts file so it can be used
again if a re-configure is triggered.
Sean Bright [Thu, 14 Oct 2021 19:38:33 +0000 (15:38 -0400)]
func_talkdetect.c: Fix logical errors in silence detection.
There are 3 separate changes here:
1. The documentation erroneously stated that the dsp_talking_threshold
argument was a number of milliseconds when it is actually an energy
level used by the DSP code to classify talking vs. silence.
2. Fixes a copy paste error in the argument handling code.
3. Don't erroneously switch to the talking state if we aren't actively
handling a frame we've classified as talking.
Patch inspired by one provided by Moritz Fain (License #6961).
Sean Bright [Thu, 7 Oct 2021 17:50:08 +0000 (13:50 -0400)]
Makefile: Use basename in a POSIX-compliant way.
If you aren't using GNU coreutils, chances are that your basename
doesn't know about the -s argument. Luckily for us, basename does what
we need it do even without the -s argument.
Naveen Albert [Mon, 24 May 2021 18:04:12 +0000 (14:04 -0400)]
chan_iax2: Add encryption for RSA authentication
Adds support for encryption to RSA-authenticated
calls. Also prevents crashes if an RSA IAX2 call
is initiated to a switch requiring encryption
but no secret is provided.
Matthew Kern [Mon, 19 Jul 2021 16:34:00 +0000 (10:34 -0600)]
res_pjsip_t38: bind UDPTL sessions like RTP
In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.
Naveen Albert [Wed, 29 Sep 2021 17:58:14 +0000 (17:58 +0000)]
app_read: Fix null pointer crash
If the terminator character is not explicitly specified
and an indications tone is used for reading a digit,
there is no null pointer check so Asterisk crashes.
This prevents null usage from occuring.
main/say.c: Support future dates with Q and q format params
The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today
Joseph Nadiv [Wed, 21 Jul 2021 21:36:03 +0000 (17:36 -0400)]
res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts
The behavior of max_contacts and remove_existing are connected. If
remove_existing is enabled, the soonest expiring contacts are removed.
This may occur when there is an unavailable contact. Similarly,
when remove_existing is not enabled, registrations from good
endpoints are rejected in favor of retaining unavailable contacts.
This commit adds a new AOR option remove_unavailable, and the effect
of this setting will depend on remove_existing. If remove_existing
is set to no, we will still remove unavailable contacts when they
exceed max_contacts, if there are any. If remove_existing is set to
yes, we will prioritize the removal of unavailable contacts before
those that are expiring soonest.
ari: Ignore invisible bridges when listing bridges.
When listing bridges we go through the ones present in
ARI, get their snapshot, turn it into JSON, and add it
to the payload we ultimately return.
An invisible "dial bridge" exists within ARI that would
also try to be added to this payload if the channel
"create" and "dial" routes were used. This would ultimately
fail due to invisible bridges having no snapshot
resulting in the listing of bridges failing.
This change makes it so that the listing of bridges
ignores invisible ones.
Sean Bright [Tue, 21 Sep 2021 14:58:09 +0000 (10:58 -0400)]
message.c: Support 'To' header override with AMI's MessageSend.
The MessageSend AMI action has been updated to allow the Destination
and the To addresses to be provided separately. This brings the
MessageSend manager command in line with the capabilities of the
MessageSend dialplan application.
Naveen Albert [Sun, 5 Sep 2021 18:11:57 +0000 (18:11 +0000)]
app_queue: Fix hint updates for included contexts
Previously, if custom hints were used with the hint:
format in app_queue, when device state changes occured,
app_queue would only do a literal string comparison of
the context used for the hint in app_queue and the context
of the hint which just changed state. This caused hints
to not update and become stale if the context associated
with the agent included the context which actually changes
state, essentially completely breaking device state for
any such agents defined in this manner.
This fix adds an additional check to ensure that included
contexts are also compared against the context which changed
state, so that the behavior is correct no matter whether the
context is specified to app_queue directly or indirectly.
Rather than stripping parameters from Content-Type headers before
comparison, first try to compare the whole string. If no match is
found, strip the parameters and try that way.
Carlos Oliva [Mon, 13 Sep 2021 15:18:11 +0000 (17:18 +0200)]
app_mp3: Force output to 16 bits in mpg123
In new mpg123 versions (since 1.26) the default output is 32 bits
Asterisk expects the output in 16 bits, so we force the output to be on 16 bits.
It will work wit new and old versions of mpg123.
Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key!
George Joseph [Tue, 14 Sep 2021 17:02:01 +0000 (11:02 -0600)]
pjproject: Add patch to fix trailing whitespace issue in rtpmap
An issue was found where a particular manufacturer's phones add a
trailing space to the end of the rtpmap attribute when specifying
a payload type that has a "param" after the format name and clock
rate. For example:
a=rtpmap:120 opus/48000/2 \r\n
Because pjmedia_sdp_attr_get_rtpmap currently takes everything after
the second '/' up to the line end as the param, the space is
included in future comparisons, which then fail if the param being
compared to doesn't also have the space.
We now use pj_scan_get() to parse the param part of rtpmap so
trailing whitespace is automatically stripped.
Naveen Albert [Thu, 2 Sep 2021 23:20:43 +0000 (23:20 +0000)]
app_stack: Include current location if branch fails
Previously, the error emitted when app_stack tries
to branch to a dialplan location that doesn't exist
has included only the information about the attempted
branch in the error log. This adds the current location
as well so users can see where the branch failed in
the logs.
Naveen Albert [Thu, 2 Sep 2021 23:57:03 +0000 (23:57 +0000)]
func_strings: Add STRBETWEEN function
Adds the STRBETWEEN function, which can be used to insert a
substring between each character in a string. For instance,
this can be used to insert pauses between DTMF tones in a
string of digits.
Naveen Albert [Fri, 3 Sep 2021 00:00:50 +0000 (00:00 +0000)]
func_env: Add DIRNAME and BASENAME functions
Adds the DIRNAME and BASENAME functions, which are
wrappers around the corresponding C library functions.
These can be used to safely and conveniently work with
file paths and names in the dialplan.
Naveen Albert [Mon, 26 Jul 2021 17:46:44 +0000 (17:46 +0000)]
func_sayfiles: Retrieve say file names
Up until now, all of the logic used to translate
arguments to the Say applications has been
directly coupled to playback, preventing other
modules from using this logic.
This refactors code in say.c and adds a SAYFILES
function that can be used to retrieve the file
names that would be played. These can then be
used in other applications or for other purposes.
Additionally, a SayMoney application and a SayOrdinal
application are added. Both SayOrdinal and SayNumber
are also expanded to support integers greater than
one billion.
Naveen Albert [Mon, 9 Aug 2021 17:41:24 +0000 (17:41 +0000)]
res_tonedetect: Tone detection module
dsp.c contains arbitrary tone detection functionality
which is currently only used for fax tone recognition.
This change makes this functionality publicly
accessible so that other modules can take advantage
of this.
Additionally, a WaitForTone and TONE_DETECT app and
function are included to allow users to do their
own tone detection operations in the dialplan.
Sean Bright [Sat, 4 Sep 2021 17:07:14 +0000 (13:07 -0400)]
term.c: Add support for extended number format terminfo files.
ncurses 6.1 introduced an extended number format for terminfo files
which the terminfo parsing in Asterisk is not able to parse. This
results in some TERM values that do support color (screen-256color on
Ubuntu 20.04 for example) to not get a color console.
Sean Bright [Tue, 7 Sep 2021 17:32:33 +0000 (13:32 -0400)]
app_voicemail.c: Ability to silence instructions if greeting is present.
There is an option to silence voicemail instructions but it does not
take into consideration if a recorded greeting exists or not. Add a
new 'S' option that does that.
When compiled without extended srtp crypto suites also disable parsing
these from received SDP. This prevents using these, as some client
implementations are not stable.
Sean Bright [Wed, 25 Aug 2021 15:21:07 +0000 (11:21 -0400)]
config_options: Handle ACO arrays correctly in generated XML docs.
There are 3 separate changes here but they are all closely related:
* Only try to set matchfield attributes on 'field' nodes
* We need to adjust how we treat the category pointer based on the
value of the category_match, to avoid memory corruption. We now
generate a regex-like string when match types other than
ACO_WHITELIST and ACO_BLACKLIST are used.
* Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
ACO_BLACKLIST_EXACT since we only have one category we need to
ignore, not two.
Mark Murawski [Tue, 31 Aug 2021 20:03:56 +0000 (16:03 -0400)]
pbx_ael: Fix crash and lockup issue regarding 'ael reload'
Currently pbx_ael does not check if a reload is currently pending
before proceeding with a reload. This can cause multiple threads to
operate at the same time on what should be mutex protected data. This
change adds protection to reloading to ensure only one ael reload is
executing at a time.
Naveen Albert [Wed, 25 Aug 2021 11:49:06 +0000 (11:49 +0000)]
app_read: Allow reading # as a digit
Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.
Naveen Albert [Wed, 25 Aug 2021 01:04:31 +0000 (01:04 +0000)]
bridge_basic: Change warning to verbose if transfer cancelled
The attended transfer feature will emit a warning if the user
cancels the transfer or the attended transfer doesn't complete
for any reason. Changes the warning to a verbose message,
since nothing is actually wrong here.
Sarah Autumn [Sun, 20 Jun 2021 04:36:34 +0000 (21:36 -0700)]
sig_analog: Changes to improve electromechanical signalling compatibility
This changeset is intended to address compatibility issues encountered
when interfacing Asterisk to electromechanical telephone switches that
implement ANI-B, ANI-C, or ANI-D.
In particular the behaviours that this impacts include:
- FGC-CAMA did not work at all when using MF signaling. Modified the
switch case block to send calls to the correct part of the
signaling-handling state machine.
- For FGC-CAMA operation, the delay between called number ST and
second wink for ANI spill has been made configurable; previously
all calls were made to wait for one full second.
- After the ANI spill, previous behavior was to require a 'ST' tone
to advance the call. This has been changed to allow 'STP' 'ST2P'
or 'ST3P' as well, for compatibility with ANI-D.
- Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.
- For calls with an ANI failure, No. 1 Crossbar switches will send
forward a single-digit failure code, with no calling number digits
and no ST pulse to terminate the spill. I've made the ANI timeout
configurable so to reduce dead air time on calls with ANI fail.
- ANI info digits configurable. Modern digital switches will send 2
digits, but ANI-B sends only a single info digit. This caused the
ANI reported by Asterisk to be misaligned.
- Changed a confusing log message to be more informative.
Andre Barbosa [Thu, 5 Aug 2021 16:55:33 +0000 (17:55 +0100)]
media_cache: Don't lock when curl the remote file
When playing a remote sound file, which is not in cache, first we need
to download it with ast_bucket_file_retrieve.
This can take a while if the remote host is slow. The current CURL
timeout is 180secs, so in extreme situations, it can take 3 minutes to
return.
Because ast_media_cache_retrieve has a lock on all function, while we
are waiting for the delayed download, Asterisk is not able to play any
more files, even the files already cached locally.
projects / thirdparty / asterisk.git / log
