Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.
Asterisk currently treats RFC2833 Digits as a single rtp payload type
with a fixed bitrate of 8K. This change would expand that to 8, 16,
24 and 32K.
This requires checking the offered rtp types for any of these bitrates
and then adding an offer for each (if configured for RFC2833.) DTMF
generation must also be changed in order to look at the current outbound
codec in order to generate appropriately timed rtp.
For cases where no outgoing audio has yet been sent prior to digit
generation, Asterisk now has a concept of a 'preferred' codec based on
offer order.
On inbound calls Asterisk will mimic the payload types of the RFC2833
digits.
On outbound calls Asterisk will choose the next free payload types starting
with 101.
UserNote: No change in configuration is required in order to enable this
feature. Endpoints configured to use RFC2833 will automatically have this
enabled. If the endpoint does not support this, it should not include it in
the SDP offer/response.
Resolves: #699
Because of the (often recursive) nature of module dependencies in
Asterisk, hot swapping a module on the fly is cumbersome if a module
is depended on by other modules. Currently, dependencies must be
popped manually by unloading dependents, unloading the module of
interest, and then loading modules again in reverse order.
To make this easier, the ability to do this recursively in certain
circumstances has been added, as an optional extension to the
"module refresh" command. If requested, Asterisk will check if a module
that has a positive usecount could be unloaded safely if anything
recursively dependent on it were unloaded. If so, it will go ahead
and unload all these modules and load them back again. This makes
hot swapping modules that provide dependencies much easier.
Resolves: #474
UserNote: In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
If the hostname field of the ast_tcptls_session_args structure is
set (which it is for websocket client connections), that hostname
will now automatically be used in an SNI TLS extension in the client
hello.
Resolves: #713
UserNote: Secure websocket client connections now send SNI in
the TLS client hello.
Commit f2f397c1a8 previously
made it possible to send Caller ID parameters to FXS stations
which, prior to that, could not be sent.
This change is complementary in that we now handle receiving
all these parameters on FXO lines and provide these up to
the dialplan, via chan_dahdi. In particular:
* If a redirecting reason is provided, the channel's redirecting
reason is set. No redirecting number is set, since there is
no parameter for this in the Caller ID protocol, but the reason
can be checked to determine if and why a call was forwarded.
* If the Call Qualifier parameter is received, the Call Qualifier
variable is set.
* Some comments have been added to explain why some of the code
is the way it is, to assist other people looking at it.
With this change, Asterisk's Caller ID implementation is now
reasonably complete for both FXS and FXO operation.
Resolves: #681
If you're tracing a large function that may call another function
multiple times in different circumstances, it can be difficult to
see from the trace output exactly which location that function
was called from. There's no good way to automatically determine
the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT
simply print out a trace line before and after the call.
The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is
that SCOPE_CALL ignores the function's return value (if any) where
SCOPE_CALL_WITH_RESULT allows you to specify the type of the
function's return value so it can be assigned to a variable.
SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT
and the "int" return type.
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
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
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
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.
If a dynamic string is created with an initial length of 0,
`ast_str_buffer(…)` will return an invalid pointer.
This was a secondary discovery when fixing #65.
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
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
* 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
Certain channel options are not set anywhere or
exposed in any way to users, making them unusable.
This exposes some of these options which make sense
for users to manipulate at runtime.
Resolves: #442
See UserNote below.
Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.
Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
603. This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).
Also extracted the XML documentation to its own file since it was
almost as large as the code itself.
UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
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.
You can now define the _TRACE_PREFIX_ macro to change the
default trace line prefix of "file:line function" to
something else. Full documentation in logger.h.
The current STIR/SHAKEN implementation is not currently usable due
to encryption issues. Rather than trying to futz with OpenSSL and
the the current code, we can take advantage of the existing
capabilities of libjwt but we first need to add it to the
third-party infrastructure already in place for jansson and
pjproject.
A few tweaks were also made to the third-party infrastructure as
a whole. The jansson "dest" install directory was renamed "dist"
to better match convention, and the third-party Makefile was updated
to clean all product directories not just the ones currently in
use.
Resolves: #349
This commit fixes crashes in JSON_DECODE() for types null, true, false
and real numbers.
In addition it ensures that a path is not deeper than 32 levels.
Also allow root object to be an array.
Add unit tests for above cases.
Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
causes the lock calls to loop over trylock in 200us intervals until
the lock is obtained and spits out log messages if it takes more
than 5 seconds. From a code perspective, the only reason they were
tied together was for logging. So... The ifdefs in lock.c were
refactored to allow DETECT_DEADLOCKS to be enabled without
also enabling DEBUG_THREADS.
Resolves: #321
UserNote: You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
To better co-exist with sounds files that may be managed by
packages, custom sound files may now be placed in
AST_DATA_DIR/sounds/custom instead of the standard
AST_DATA_DIR/sounds/<lang> directory. If the new
"sounds_search_custom_dir" option in asterisk.conf is set
to "true", asterisk will search the custom directory for sounds
files before searching the standard directory. For performance
reasons, the "sounds_search_custom_dir" defaults to "false".
Resolves: #315
UserNote: A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/<lang> directory.
The previous behavior of make_buildopts_h was to not add the
non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS,
REF_DEBUG, etc. to the buildopts.h file because "it caused
ccache to invalidate files and extended compile times". They're
only defined by passing them on the gcc command line with '-D'
options. In practice, including them in the include file rarely
causes any impact because the only time ccache cares is if you
actually change an option so the hit occurrs only once after
you change it.
OK so why would we want to include them? Many IDEs follow the
include files to resolve defines and if the options aren't in an
include file, it can cause the IDE to mark blocks of "ifdeffed"
code as unused when they're really not.
So...
* Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H
which tells make_buildopts_h to include the non-ABI-breaking
flags in buildopts.h as well as the ABI-breaking ones. The default
is disabled to preserve current behavior. As before though,
only the ABI-breaking flags appear in AST_BUILDOPTS and only
those are used to calculate AST_BUILDOPT_SUM.
A new AST_BUILDOPT_ALL define was created to capture all of the
flags.
* make_version_c was streamlined to use buildopts.h and also to
create asterisk_build_opts_all[] and ast_get_build_opts_all(void)
* "core show settings" now shows both AST_BUILDOPTS and
AST_BUILDOPTS_ALL.
UserNote: The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.
Resolves: #71
UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
Fixes#221
UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
The ast_app_getdata() and ast_app_getdata_terminator() declarations
in app.h were changed recently to return enum ast_getdata_result
(which is how they were defined in app.c). The existing
declaration of ast_getdata_result in app.h was about 1000 lines
after those functions however so under certain circumstances,
a "use before declaration" error was thrown by the compiler.
The declaration of the enum was therefore moved to before those
functions.
Resolves: #200
Added two new functions (ast_sip_session_get_dialog and
ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the
pjsip_inv_state respectively from the pjsip_inv_session on the
ast_sip_session struct. This is due to pjproject adding a new field to
the pjsip_inv_session struct that caused crashes when trying to access
fields that were no longer where they were expected to be if a module
was compiled against a different version of pjproject.
Resolves: #145
These were uncovered when trying to run `bootstrap.sh` with Autoconf
2.71:
* AC_CONFIG_HEADER() is deprecated in favor of AC_CONFIG_HEADERS().
* AC_HEADER_TIME is obsolete.
* $as_echo is deprecated in favor of AS_ECHO() which requires an update
to ax_pthread.m4.
Note that the generated artifacts in this commit are from Autoconf 2.69.
Resolves#139
* gcc 13 is now catching when a function is declared as returning
an enum but defined as returning an int or vice versa. Fixed
a few in app.h, loader.c, stasis_message.c.
* gcc 13 is also now (incorrectly) complaining of dangling pointers
when assigning a pointer to a local char array to a char *. Had
to change that to an ast_alloca.
Resolves: #155
The Caller ID generation routine currently is hardcoded
to always use the system time zone. This makes it possible
to optionally specify any TZ-format time zone.
Resolves: #98
ASTERISK-30330
With this change, session modifications in the early media state are
possible if the SDP was sent reliably and confirmed by a PRACK. For
details, see RFC 6337, escpecially section 3.2.
Resolves: #73
The existing res_pjsip_pubsub APIs are somewhat limited in
what they can do. This adds a few API extensions that make
it possible for PJSIP pubsub modules to implement richer
features than is currently possible.
* Allow pubsub modules to get a handle to pjsip_rx_data on subscription
* Allow pubsub modules to run a callback when a subscription is renewed
* Allow pubsub modules to run a callback for outgoing NOTIFYs, with
a handle to the tdata, so that modules can append their own headers
to the NOTIFYs
This change does not add any features directly, but makes possible
several new features that will be added in future changes.
Resolves: #81
ASTERISK-30485 #close
Master-Only: True
Deprecate `ast_gethostbyname()` in favor of `ast_sockaddr_resolve()` and
`ast_sockaddr_resolve_first_af()`. `ast_gethostbyname()` has not been
used by any in-tree code since 2021.
This function will be removed entirely in Asterisk 23.
Resolves: #78
UpgradeNote: ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use `ast_sockaddr_resolve()` and
`ast_sockaddr_resolve_first_af()`.
The current AST_CEL_LOCAL_OPTIMIZE event is and has been
triggered on a local optimization end to serve as a flag
indicating the event occurred. This change adds a second
AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.
Resolves: #52
UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
* 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
When using mediasec, requests sent after a 401 must still contain the
Security-Client header according to
draft-dawes-sipcore-mediasec-parameter.
Resolves: #48
The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.
RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.
To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.
Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.
ASTERISK-30407 #close
Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2
Adds PJSIP as a supported technology to DUNDi.
To facilitate this, we now allow an endpoint to be specified
for outgoing PJSIP calls. We also allow users to force a specific
channel technology for outgoing SIP-protocol calls.
ASTERISK-28109 #close
ASTERISK-28233 #close
Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf
While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.
This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute. This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.
ASTERISK-30464
Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98
* Added a new function ast_utf8_replace_invalid_chars() to
utf8.c that copies a string replacing any invalid UTF-8
sequences with the Unicode specified U+FFFD replacement
character. For example: "abc\xffdef" becomes "abc\uFFFDdef".
Any UTF-8 compliant implementation will show that character
as a � character.
* Updated res_pjsip:set_id_from_hdr() to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
* Updated stasis_channels:ast_channel_publish_varset to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
ASTERISK-27830
Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8
Removed multiple patches.
Code chages in res_pjsip_pubsub due to changes in evsub.
Pjsip now calls on_evsub_state() before on_rx_refresh(),
so the sub tree deletion that used to take place in
on_evsub_state() now must take place in on_rx_refresh().
Additionally, pjsip now requires that you send the NOTIFY
from within on_rx_refresh(), otherwise it will assert
when going to send the 200 OK. The idea is that it will
look for this NOTIFY and cache it until after sending the
response in order to deal with the self-imposed message
mis-order. Asterisk previously dealt with this by pushing
the NOTIFY in on_rx_refresh(), but pjsip now forces us
to use it's method.
Changes were required to configure in order to detect
which way pjsip handles this as the two are not
compatible for the reasons mentioned above.
A corresponding change in testsuite is required in order
to deal with the small interal timing changes caused by
moving the NOTIFY send.
ASTERISK-30325
Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f
Adds the overlap_context option, which can be used
to explicitly specify a context to use for overlap
dialing extension matches, rather than forcibly
using the context configured for the endpoint.
ASTERISK-30262 #close
Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db
For most modules that interacted with app_macro, this change is limited
to no longer looking for the current context from the macrocontext when
set. Additionally, the following modules are impacted:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs a re-write
app_queue - can no longer a macro on the called party's channel.
Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected line or
redirection macro options
options - stdexten is deprecated to gosub as the default and only
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
ASTERISK-30304
Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6
Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).
ASTERISK-30319 #close
Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645
-----------------
This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures. The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.
ASTERISK-30391
-----------------
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038
json.h contains macros to get a string and an integer
from a JSON object. However, the macro to do this for
JSON reals is missing. This adds that.
ASTERISK-30361 #close
Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
ASTERISK-30280
Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a
Currently, there is no Caller ID available to us when
checking for an extension match when handling INVITEs.
As a result, extension patterns that depend on the Caller ID
are not matched and calls may be incorrectly rejected.
The Caller ID is not available because the supplement that
adds Caller ID to the session does not execute until after
this check. Supplement callbacks cannot yet be executed
at this point since the session is not yet in the appropriate
state.
To fix this without impacting existing behavior, the Caller ID
number is now retrieved before attempting to pattern match.
This ensures pattern matching works correctly and there is
no behavior change to the way supplements are called.
ASTERISK-28767 #close
Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7
When ast_stream_and_wait returns an error (for example, when attempting
to stream to a channel after hangup) the stream is not closed, and
callers typically do not check the return code. This results in leaking
file descriptors, leading to resource exhaustion.
This change ensures that the stream is closed in case of error.
ASTERISK-30198 #close
Reported-by: Julien Alie
Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803
chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).
This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.
* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)
The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.
The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.
Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.
ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>
Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333
This file includes some doxygen comments referencing
ast_format_set. This is an obsolete API that was
removed years back, but documentation was not fully
updated to reflect that. These examples are
updated to the current way of doing things
(using the format cache).
ASTERISK-30327 #close
Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7
Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.
ASTERISK-30176
Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404
(cherry picked from commit 81f10e847e)
It was possible for a module that registered for transport monitor
events to pass in a pjsip_transport that had already been freed.
This caused pjsip_transport_events to crash when looking up the
monitor for the transport. The fix is a two pronged approach.
1. We now increment the reference count on pjsip_transports when we
create monitors for them, then decrement the count when the
transport is going to be destroyed.
2. There are now APIs to register and unregister monitor callbacks
by "transport key" which is a string concatenation of the remote ip
address and port. This way the module needing to monitor the
transport doesn't have to hold on to the transport object itself to
unregister. It just has to save the transport_key.
* Added the pjsip_transport reference increment and decrement.
* Changed the internal transport monitor container key from the
transport->obj_name (which may not be unique anyway) to the
transport_key.
* Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that
fills a buffer with the transport_key using a passed-in
pjsip_transport.
* Added the following functions:
ast_sip_transport_monitor_register_key
ast_sip_transport_monitor_register_replace_key
ast_sip_transport_monitor_unregister_key
and marked their non-key counterparts as deprecated.
* Updated res_pjsip_pubsub and res_pjsip_outbound_register to use
the new "key" monitor functions.
NOTE: res_pjsip_registrar also uses the transport monitor
functionality but doesn't have a persistent object other than
contact to store a transport key. At this time, it continues to
use the non-key monitor functions.
ASTERISK-30244
Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b
(cherry picked from commit 7684c9e907)
If multiple codecs are available for the same
resource and the translation costs between
multiple codecs are the same, ties are
currently broken arbitrarily, which means a
lower quality codec would be used. This forces
Asterisk to explicitly use the higher quality
codec, ceteris paribus.
ASTERISK-29455
Change-Id: I4b7297e1baca7aac14fe4a3c7538e18e2dbe9fd6
Currently chan_pjsip on receiving a re-INVITE without SDP will only
return the codecs that are previously negotiated and not offering
all enabled codecs.
This causes interoperability issues with different equipment (e.g.
from Cisco) for some of our customers and probably also in other
scenarios involving 3PCC infrastructure.
According to RFC 3261, section 14.2 we SHOULD return all codecs
on a re-INVITE without SDP
The PR proposes a new parameter to configure this behaviour:
all_codecs_on_empty_reinvite. It includes the code, documentation,
alembic migrations, CHANGES file and example configuration additions.
ASTERISK-30193 #close
Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148
Add enum to allow setting optional direction. If set to only one
direction, only feed matching-direction frames to the associated
slin factory.
This prevents mangling the transcoder on non-mixed frames when the
READ and WRITE frames would have otherwise required it. Also
removes the need to mute or discard the un-wanted frames as they
are no longer added in the first place.
res_stasis_snoop is changed to use this addition to set direction
on audiohook based on spy direction.
If no direction is set, the ast_audiohook_init will init this enum
to BOTH which maintains existing functionality.
ASTERISK-30252
Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb
Allows bridging, parking, and dial messages to be globally
ignored for all CDRs such that only a single CDR record
is generated per channel.
This is useful when CDRs should endure for the lifetime of
an entire channel and bridging and dial updates in the
dialplan should not result in multiple CDR records being
created for the call. With the ignore bridging option,
bridging changes have no impact on the channel's CDRs.
With the ignore dial state option, multiple Dials and their
outcomes have no impact on the channel's CDRs. The
last disposition on the channel is preserved in the CDR,
so the actual disposition of the call remains available.
These two options can reduce the amount of "CDR hacks" that
have hitherto been necessary to ensure that CDR was not
"spoiled" by these messages if that was undesired, such as
putting a dummy optimization-disabled local channel between
the caller and the actual call and putting the CDR on the channel
in the middle to ensure that CDR would persist for the entire
call and properly record start, answer, and end times.
Enabling these options is desirable when calls correspond
to the entire lifetime of channels and the CDR should
reflect that.
Current default behavior remains unchanged.
ASTERISK-30091 #close
Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832
This patch adds support for mediasec SIP headers and SDP attributes.
These are defined in RFC 3329, 3GPP TS 24.229 and
draft-dawes-sipcore-mediasec-parameter. The new features are
implemented so that a backbone for RFC 3329 is present to streamline
future work on RFC 3329.
With this patch, Asterisk can communicate with Deutsche Telekom trunks
which require these fields.
ASTERISK-30032
Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2
This patch adds a new option to the 100rel parameter for pjsip
endpoints called "peer_supported". When an endpoint with this option
receives an incoming request and the request indicated support for the
100rel extension, then Asterisk will send 1xx responses reliably. If
the request did not indicate 100rel support, Asterisk sends 1xx
responses normally.
ASTERISK-30158
Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad
Fixed a segfault caused by var_list_from_loc_info() encountering
an empty location info element.
Fixed an issue in ast_strsep() where a value with only whitespace
wasn't being preserved.
Fixed an issue in ast_variable_list_from_quoted_string() where
an empty value was considered a failure.
ASTERISK-30215
Reported by: Dan Cropp
Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a
This change allows TEL URI requests to come through for basic calls. The
allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To
headers will now allow TEL URIs, as well as the request URI.
Support is only for TEL URIs present in traffic from a remote party.
Asterisk does not generate any TEL URIs on its own.
ASTERISK-26894
Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a
We're validating the following functionality:
encrypting a block of data with RSA
decrypting a block of data with RSA
signing a block of data with RSA
verifying a signature with RSA
encrypting a block of data with AES-ECB
encrypting a block of data with AES-ECB
as well as accessing test keys from the keystore.
ASTERISK-30045 #close
Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d
Use const char for char arguments to
pbx_substitute_variables_helper_full_location
that can do so (context and exten).
ASTERISK-30209 #close
Change-Id: I001357177e9c3dca2b2b4eebc5650c1095b3da6f
Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.
Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.
Added a few missing parameters to the ones allowed for writing
with GEOLOC_PROFILE.
Fixed a bug where calling GEOLOC_PROFILE to read a parameter
might actually update the profile object.
Cleaned up XML documentation a bit.
ASTERISK-30190
Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c
You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles. This is
mutually exclusive with setting location_reference on the
profile.
Updated appdocsxml.dtd to allow xi:include in a configObject
element. This makes it easier to link to complete configOptions
in another object. This is used to add the above fields to the
profile object without having to maintain the option descriptions
in two places.
ASTERISK-30185
Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569
Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.
Fixed a possible SEGV if a sub-parameter value didn't have a
value.
ASTERISK-30177
Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd
If "core show channels" is run before startup has completed, it
is possible for bad ao2 refs to occur because the system is not
yet fully initialized. This will lead to an assertion failing.
To prevent this, initialization of CLI builtins is moved to be
later along in the main load sequence. Core CLI commands are
loaded at the same time, but channel-related commands are loaded
later on.
ASTERISK-29846 #close
Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6
This change adds support using the pjsip_tls_transport_restart
function for reloading the TLS certificate and key, if the filenames
remain unchanged. This is useful for Let's Encrypt and other
situations. Note that no restart of the transport will occur if
the certificate and key remain unchanged.
ASTERISK-30186
Change-Id: I9bc95a6bf791830a9491ad9fa43c17d4010028d0
The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set()
for us, so no need to declare it separately.
Change-Id: I4813a884ada475ddc62bca480bceb4a53b3ec59a
Adds additional control options over the transfer
feature functionality to give users more control
in how the transfer feature sounds and works.
First, the "transfer" sound that plays when a transfer is
initiated can now be customized by the user in
features.conf, just as with the other transfer sounds.
Secondly, the user can now specify the transfer extension
in advance by using the TRANSFER_EXTEN variable. If
a valid extension is contained in this variable, the call
will automatically be transferred to this destination.
Otherwise, it will fall back to collecting the extension
from the user as is always done now.
ASTERISK-29899 #close
Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3
Fixes a few coding guideline violations:
* Use of C99 comments
* Opening brace on same line as function prototype
ASTERISK-30163 #close
Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7
* Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
reflect it's purpose.
* Added the config option for 'allow_routing_use' which
sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
* Removed the 'profile' argument.
* Automatically create a profile if it doesn't exist.
* Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.
ASTERISK-30167
Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0
There are several things wrong with analog Caller ID
handling that are fixed by this commit:
callerid.c's Caller ID generation function contains the
logic to use the presentation to properly send the proper
Caller ID. However, currently, DAHDI does not pass any
presentation information to the Caller ID module, which
means that presentation is completely ignored on all calls.
This means that lines could be getting Caller ID information
they aren't supposed to.
Part of the reason this has been obscured is because the
simple switch logic for handling the built in *67 and *82
is completely wrong. Rather than modifying the presentation
for the call accordingly (which is what it's supposed to do),
it simply blanks out the Caller ID or fills it in. This is
wrong, so wrong that it makes a mockery of the specification.
Additionally, it would leave to the "UNAVAILABLE" disposition
being used for Caller ID generation as opposed to the "PRIVATE"
disposition that it should have been using. This is now fixed
to only update the presentation and not modify the number and
name, so that the simple switch *67/*82 work correctly.
Next, sig_analog currently only copies over the name and number,
nothing else, when it is filling in a duplicated caller id
structure. Thus, we also now copy over the presentation
information so that is available for the Caller ID spill.
Additionally, this meant that "valid" was implicitly 0,
and as such presentation would always fail to "Unavailable".
The validity is therefore also copied over so it can be used
by ast_party_id_presentation.
As part of this fix, new API is added so that all the relevant
Caller ID information can be passed in to the Caller ID generation
functions. Parameters that are also completely missing from the
Caller ID spill have also been added, to enhance the compatibility,
correctness, and completeness of the Asterisk Caller ID implementation.
ASTERISK-29991 #close
Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15
This commit adds res_pjsip_geolocation which gives chan_pjsip
the ability to use the core geolocation capabilities.
This commit message is intentionally short because this isn't
a simple capability. See the documentation at
https://wiki.asterisk.org/wiki/display/AST/Geolocation
for more information.
THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON
USER FEEDBACK!
ASTERISK-30128
Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9
This commit adds res_geolocation which creates the core capabilities
to manipulate Geolocation information on SIP INVITEs.
An upcoming commit will add res_pjsip_geolocation which will
allow the capabilities to be used with the pjsip channel driver.
This commit message is intentionally short because this isn't
a simple capability. See the documentation at
https://wiki.asterisk.org/wiki/display/AST/Geolocation
for more information.
THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON
USER FEEDBACK!
ASTERISK-30127
Change-Id: Ibfde963121b1ecf57fd98ee7060c4f0808416303
* Added ast_variable_list_from_quoted_string()
Parse a quoted string into an ast_variable list.
* Added ast_str_substitute_variables_full2()
Perform variable/function/expression substitution on an ast_str.
* Added ast_strsep_quoted()
Like ast_strsep except you can specify a specific quote character.
Also added unit test.
* Added ast_xml_find_child_element()
Find a direct child element by name.
* Added ast_xml_doc_dump_memory()
Dump the specified document to a buffer
* ast_datastore_free() now checks for a NULL datastore
before attempting to destroy it.
Change-Id: I5dcefed2f5f93a109e8b489e18d80d42e45244ec
Currently, if using the CLI to delete a DB entry,
"Database entry removed" is always returned,
regardless of whether or not the entry actually
existed in the first place. This meant that users
were never told if entries did not exist.
The same issue occurs if trying to delete a DB key
using AMI.
To address this, new API is added that is more stringent
in deleting values from AstDB, which will not return
success if the value did not exist in the first place,
and will print out specific error details if available.
ASTERISK-30001 #close
Change-Id: Ic84e3eddcd66c7a6ed7fea91cdfd402568378b18
Rightly the use of wildcards in certificates is disallowed in accordance
with RFC5922. However, RFC2818 does make some allowances with regards to
their use when using subject alt names with DNS name types.
As such this patch creates a new setting for TLS transports called
'allow_wildcard_certs', which when it and 'verify_server' are both enabled
allows DNS name types, as well as the common name that start with '*.'
to match as a wildcard.
For instance: *.example.com
will match for: foo.example.com
Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc...
And the starting wildcard only matches for a single level.
For instance: *.example.com
will NOT match for: foo.bar.example.com
The new setting is disabled by default.
ASTERISK-30072 #close
Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4
Finding an application and executing it if found is
a common task throughout Asterisk. This adds a helper
function around pbx_exec to do this, to eliminate
redundant code and make it easier for modules to
substitute variables and execute applications by name.
ASTERISK-30061 #close
Change-Id: Ifee4d2825df7545fb515d763d393065675140c84
Sean Bright
George Joseph
Mike Bradeen
Naveen Albert
Sperl Viktor
Joshua Elson
Joshua C. Colp
Brad Smith
sungtae kim
Bastian Triller
Maximilian Fridrich
Ben Ford
InterLinked1
Sean Bright
Igor Goncharovsky
George Joseph
Peter Fern
Michael Kuron
Henning Westerholt
Philip Prindeville
Kevin Harwell