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
Adds support for detecting audible ringback tone
to the TONE_DETECT function using the p option.
ASTERISK-30254 #close
Change-Id: Ie2329ff245248768367d26749c285fbe823f6414
"fname" is passed in as a const char *, but strstr() mangles that
into a char *, and we were attempting to modify the string in place.
This is an unwanted (and undocumented) side-effect.
ASTERISK-30213
Change-Id: Ifa36d352aafeb7f9beec3f746332865c7d21e629
Also added a note to the geolocation.conf.sample file
and added a README to the res/res_geolocation/wiki
directory.
Change-Id: I89c3c5db8c0701b33127993622d5e4f904bddfbc
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
Avoid crashing by skipping invisible bridges and checking the
snapshot for a null pointer. In effect this is how the bridges
are enumerated in res/ari/resource_bridges.c already.
ASTERISK-30239
ASTERISK-30237
Change-Id: I58ef9f44036feded5966b5fc70ae754f8182883d
If geolocation is not in use for an endpoint, the NOTICE
log level is currently spammed with messages about this,
even though nothing is wrong and these messages provide
no real value. These log messages are therefore changed
to debugs.
ASTERISK-30241 #close
Change-Id: I656b355d812f67cc0f0fdf09b00b0e1458598bb4
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
Adding user=phone to local-side uri's when user_eq_phone=yes is set for
an endpoint. Previously this would only add the header to the To and R-URI.
ASTERISK-30178
Change-Id: Id3bfb5d225d762e7d2668c023fe09e4541ae8600
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 adds an option, answeredonly, that will prevent music on
hold on channels that are not answered.
ASTERISK-30135
Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08
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
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
The trigger to perform outgoing geolocation processing is the
presence of a geoloc_outgoing_call_profile on an endpoint. This
is intentional so as to not leak location information to
destinations that shouldn't receive it. In a totally dynamic
configuration scenario however, there may not be any profiles
defined in geolocation.conf. This makes it impossible to do
outgoing processing without defining a "dummy" profile in the
config file.
This commit adds 4 built-in profiles:
"<prefer_config>"
"<discard_config>"
"<prefer_incoming>"
"<discard_incoming>"
The profiles are empty except for having their precedence
set and can be set on an endpoint to allow processing without
entries in geolocation.conf. "<discard_config>" is actually the
best one to use in this situation.
ASTERISK-30182
Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a
When producing an outgoing SDP we iterate through the configured
formats and produce SDP information. It is possible for some
configured formats to not have SDP information available. If this
is the case we skip over them to allow the SDP to still be
produced.
ASTERISK-29185
Change-Id: I3e37569aa4ca341260e6ca5904dc2f75e46a1749
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
Fixes two typos that cause fax detection to not work.
One refers to the wrong frame variable, and the other
refers to the subclass.integer instead of the frametype
as it should.
ASTERISK-30192 #close
Change-Id: I7b35fdb7bcf25a29a212eee37c20812c64ab3ef1
Set termination state to old subscriptions to prevent queueing and sending
NOTIFY messages on exten/device state changes.
Postpone destruction of old subscriptions until all already queued tasks
that may be using old subscriptions have completed.
ASTERISK-29906
Change-Id: I96582aad3a26515ca73a8460ee6756f56f6ba23b
* 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 a handful of files in the tree that
reference an SVN link for the coding guidelines.
This removes these because the links are dead
and the vast majority of source files do not
contain these links, so this is more consistent.
app_skel still maintains an (up to date) link
to the coding guidelines.
ASTERISK-30159 #close
Change-Id: I35bbb20f66982e98099cff3029ede20091ffdac7
Adjusts some logging levels to be more or less important,
that is more prominent when actual problems occur and less
prominent for less noteworthy things.
ASTERISK-30153 #close
Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b
Move the call to ast_sip_location_prune_boot_contacts() *after* the call
to ast_res_pjsip_init_options_handling() so that
res/res_pjsip/pjsip_options.c is informed about the contact deletion and
updates its sip_options_contact_statuses list. This allows for an AMI
event to be sent by res/res_pjsip/pjsip_options.c if the endpoint
registers again from the same remote address and port (i.e., same URI)
as used before the Asterisk restart.
ASTERISK-30109
Reported-by: Michael Neuhauser
Change-Id: I1ba4478019e4931a7085f62708d9b66837e901a8
line 196: loc_src = '\0';
should have been
line 196: *loc_src = '\0';
The issue was caught by the gcc optimizer complaining that
loc_src had a zero length because the pointer itself was being
set to NULL instead of the _contents_ of the pointer being set
to the NULL terminator.
ASTERISK-30138
Reported-by: Sean Bright
Change-Id: Id247be113cc8510f043ca053d5b4f5f3d32acd29
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
A sporadic test failure was happening when executing the AEAP
Websocket transport tests. It was originally thought this was
due to things not getting cleaned up fast enough, but upon further
investigation I determined the underlying cause was poll()
getting interrupted and this not being handled in all places.
This change adds EINTR and EAGAIN handling to the Websocket
client connect code as well as the AEAP Websocket transport code.
If either occur then the code will just go back to waiting
for data.
The originally disabled failure test case has also been
re-enabled.
ASTERISK-30099
Change-Id: I1711a331ecf5d35cd542911dc6aaa9acf1e172ad
Adds a CLI command similar to "dialplan eval function" except for
applications: "dialplan exec application", useful for quickly
testing certain application behavior directly from the CLI
without writing any dialplan.
ASTERISK-30062 #close
Change-Id: I42e9fa9b60746c21450d40f99a026d48d2486dde
These new functions allow retrieving information from headers on 200 OK
INVITE response.
ASTERISK-29999
Change-Id: I264a610a9333359297a0825feb29a1bb4f4ad144
Switched res_pjsip_outbound_registration.so dep to optional. Added
module loaded check before using it.
ASTERISK-30101 #close
Change-Id: Ia34f1684d984e821fbdd4de8911f930337703666
Microsoft recently began rejecting all requests for
ICS calendars on Office 365 with 400 errors if
the request doesn't contain a user agent. See:
https://docs.microsoft.com/en-us/answers/questions/883904/34the-remote-server-returned-an-error-400-bad-requ.html
Accordingly, we now send a user agent on requests for
ICS files so that requests to Office 365 will work as
they did before.
ASTERISK-30106
Change-Id: Ie9dcaef12ae8adf37533c684499eb11005fac8f7
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
An m option to Park and ParkAndAnnounce now allows
specifying a music on hold class override.
ASTERISK-30087
Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e
Currently, PJSIP will randomly wait up to 10 seconds for each
outbound registration's initial attempt. The reason for this
is to avoid having all outbound registrations attempt to register
simultaneously.
This can create limitations with the test suite where we need to
be able to receive inbound calls potentially within 10 seconds of
starting up. For instance, we might register to another server
and then try to receive a call through the registration, but if
the registration hasn't happened yet, this will fail, and hence
this inconsistent behavior can cause tests to fail. Ultimately,
this requires a smaller random value because there may be no good
reason to wait for up to 10 seconds in these circumstances.
To address this, a new config option is introduced which makes this
maximum delay configurable. This allows, for instance, this to be
set to a very small value in test systems to ensure that registrations
happen immediately without an unnecessary delay, and can be used more
generally to control how "tight" the initial outbound registrations
are.
ASTERISK-29965 #close
Change-Id: Iab989a8e94323e645f3a21cbb6082287c7b2f3fd
When a pjsip endpoint is defined with timers=always, this has been a
functional noop. This patch correctly sets the feature bitmap to both
enable support for session timers and to enable them even when the
endpoint itself does not request or support timers.
ASTERISK-29603
Reported-By: Ray Crumrine
Change-Id: I8b5eeaa9ec7f50cc6d96dd34c2b4aa9c53fb5440
If there is scheduled notification, we must delete it
to avoid using destroyed subscriptions.
ASTERISK-29906
Change-Id: I1c644e5e15a8fe43eed8e4f9112f113cbf87a40f
res_calendar will trigger an assertion currently
if the ending time is calculated to be in the past.
Unlike the reminder and start times, however, there
is currently no check to catch non-positive times
and set them to 1. As a result, if we get a negative
value by happenstance, this can cause a crash.
To prevent the assertion from begin triggered, we now
use the same logic as the reminder and start events
to catch this issue before it can cause a problem.
ASTERISK-29981 #close
Change-Id: Idfb3204d195f350d2575fb4bc72a54a597d6e93c
Emits a warning if the user has requested a parking spot that
is out of bounds for the requested parking lot.
ASTERISK-30086
Change-Id: I1080371e4f63e94724455003753014fbd3f95fbf
The change "Add LOCAL/REMOTE tags in dialog-info+xml" set both "local"
Identity Element URI and Target Element URI to the same value -
the channel Caller Number.
For Identity Element it's ok to set as Caller ID.
But Local Target URI should be set as local URI.
In this case the Local Target URI can be used for Directed Call Pickup
by Polycom ip-phones (parameter useLocalTargetUriforLegacyPickup).
Also XML sanitized Display names.
ASTERISK-24601
Change-Id: If130a2f2f3b2339b14dca0ec0ebeea3a87b34343
Agi commnad exec can now evaluate dialplan functions and
variables if variable AGIEXECFULL is set to yes. this can
be useful when executing Playback or Read from agi.
ASTERISK-30058 #close
Change-Id: I669991f540496e7bddd096fec82b52c083036832
This change exposes the channel driver's unique id (i.e. the Call-ID
for chan_sip/chan_pjsip based channels) to ARI channel resources
as `protocol_id`.
ASTERISK-30027
Reported by: Moritz Fain
Tested by: Moritz Fain
Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87
As part of PJSIP 2.11 a behavior change was done to require
a matching remote hostname on an established transport for
secure transports. Since the Websocket transport is considered
a secure transport this caused the existing connection to not
be found and used.
We now set the remote hostname and the transport can be found.
ASTERISK-30065
Change-Id: Ia1cdef33e1411f927985b4b852c95e163c080e94
This is needed to be able to restore it in REGISTER responses,
otherwise the client won't be able to find the contact it created.
ASTERISK-30042
Change-Id: I0c5823918199acf09246b3b206fbde66773688f6
Adjusts the pjsip show registration(s) commands to show
the amount of seconds remaining until a registration
expires.
ASTERISK-29845 #close
Change-Id: Ic4fea15a1a1056c424416def49d1ca8e776c0483
Most issues were in stringfields and had to do with comparing
a pointer to an constant/interned string with NULL. Since the
string was a constant, a pointer to it could never be NULL so
the comparison was always "true". gcc now complains about that.
There were also a few issues where determining if there was
enough space for a memcpy or s(n)printf which were fixed
by defining some of the involved variables as "volatile".
There were also a few other miscellaneous fixes.
ASTERISK-30044
Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570
Adds version information for applications, functions,
and manager events/actions.
This is not completely exhaustive by any means but
covers most new things added that have release
versioning information in the issue tracker.
ASTERISK-29940 #close
Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131
added new global config option "allow_sending_180_after_183"
that if enabled will preserve 180 after a 183
ASTERISK-29842
Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18
Add framework to connect to, and read and write protocol based
messages from and to an external application using an Asterisk
External Application Protocol (AEAP). This has been divided into
several abstractions:
1. transport - base communication layer (currently websocket only)
2. message - AEAP description and data (currently JSON only)
3. transaction - links/binds requests and responses
4. aeap - transport, message, and transaction handler/manager
This patch also adds an AEAP implementation for speech to text.
Existing speech API callbacks for speech to text have been completed
making it possible for Asterisk to connect to a configured external
translator service and provide audio for STT. Results can also be
received from the external translator, and made available as speech
results in Asterisk.
Unit tests have also been created that test the AEAP framework, and
also the speech to text implementation.
ASTERISK-29726 #close
Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2
There was an issue with the conditional where STIR/SHAKEN would be
enabled even when not configured. It has been changed to ensure that if
a profile does not exist and stir_shaken is not set in pjsip.conf, then
the conditional will return from the function without performing
STIR/SHAKEN operations.
ASTERISK-30024
Change-Id: I41286a3d35b033ccbfbe4129427a62cb793a86e6
The async_operations setting on a transport configures how
many simultaneous incoming packets the transport can handle
when multiple threads are polling and waiting on the transport.
As we only use a single thread this was needlessly creating
incoming packets when set to a non-default value, wasting memory.
ASTERISK-30006
Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f
Adds a new configuration option, stir_shaken_profile, in pjsip.conf that
can be specified on a per endpoint basis. This option will reference a
stir_shaken_profile that can be configured in stir_shaken.conf. The type
of this option must be 'profile'. The stir_shaken option can be
specified on this object with the same values as before (attest, verify,
on), but it cannot be off since having the profile itself implies wanting
STIR/SHAKEN support. You can also specify an ACL from acl.conf (along
with permit and deny lines in the object itself) that will be used to
limit what interfaces Asterisk will attempt to retrieve information from
when reading the Identity header.
ASTERISK-29476
Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406
Put checks in place to limit how much we will actually download, as well
as a check for the data we receive at the start to ensure it begins with
what we would expect a certificate to begin with.
ASTERISK-29872
Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46
Removes some leftover build and config references to
modules that have since been removed from Asterisk.
ASTERISK-29935 #close
Change-Id: Iaefc73a23f4b2de3c6c14d928050135b6d0ef6af
When adding headers to an outgoing request the headers were cloned using
the dialog's pool when they should have been cloned using tdata's pool.
Under certain circumstances it was possible for the dialog object, and
its pool to be freed while tdata is still active and available. Thus the
cloned header "disappeared", and when tdata tried to later access it a
crash would occur.
This patch makes it so all added headers are cloned appropriately using
tdata's pool.
ASTERISK-29411 #close
ASTERISK-29535 #close
Change-Id: I9852025b5ee93ce1c038209150ee9dba1e0767c5
There is work going on to update our OpenSSL usage to avoid the
deprecated functions but in the meantime make it possible to compile
in devmode.
Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726
Change RTP timer behavior for detecting RTP only after two-way
SDP channel establishment. Ignore detecting after receiving 183
with SDP or while direct media is used.
Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout
and rtpholdtimeout options in chan_sip.
ASTERISK-26689 #close
ASTERISK-29929 #close
Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb
Treat time_t's as entirely unique and use the POSIX API's for
converting to/from strings.
Lastly, a 64-bit integer formats as 20 digits at most in base10.
Don't need to have any 100 byte buffers to hold that.
ASTERISK-29674 #close
Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56
When asterisk generates the RLMI part of NOTIFY request,
the asterisk uses the local contact uri instead of the URI to which
the SUBSCRIBE request is sent.
Because of this mismatch some IP phones (for example Cisco 5XX) ignore
this list.
According
https://datatracker.ietf.org/doc/html/rfc4662#section-5.2
The first mandatory <list> attribute is "uri", which contains the uri
that corresponds to the list. Typically, this is the URI to which
the SUBSCRIBE request was sent.
https://datatracker.ietf.org/doc/html/rfc4662#section-5.3
The "uri" attribute identifies the resource to which the <resource>
element corresponds. Typically, this will be a SIP URI that, if
subscribed to, would return the state of the resource.
This patch makes asterisk to generate URI using SUBSCRIBE request URI.
ASTERISK-29961 #close
Change-Id: I1fcfc08fd589677f40608c59a4e143c45ee05f6c
Using the length of a file found on the filesystem rather than the
file being requested could result in filenames whose names are
substrings of another to be erroneously matched.
We now ensure a complete comparison before returning a positive
result.
ASTERISK-29960 #close
Change-Id: Id3ffc77681b9b75b8569062f3d952a128a21c71a
This patch makes the Resource List Subscriptions (RLS) dynamic.
The asterisk updates the current subscriptions to reflect the changes
to the list on the subscriptions refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.
ASTERISK-29906 #close
Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3
The XML documentation for the SET MUSIC AGI
command is invalid, as the parameter does not
have a name and the on/off enum options for
the on/off argument are listed separately, which
is incorrect. The cumulative effect of these currently
is that the Asterisk Wiki documentation for SET MUSIC
is broken and external documentation generators crash
on SET MUSIC due to the malformed documentation.
These issues are corrected so that the documentation
can be successfully parsed as with other similar AGI
commands.
ASTERISK-29939 #close
ASTERISK-28891 #close
Change-Id: I8c3d59897531bcbc401cbc7b00c9e2829dcb35f8
Omit "unsupported column type 'text'" warning in logs while
using text-type column in the PgSQL backend.
ASTERISK-29924 #close
Change-Id: I48061a7d469426859670db07f1ed8af1eb814712
Added the missing xml-stylesheet and Xinclude namespace
declarations in pjsip_config.xml and pjsip_manager.xml.
Updated make_xml_documentation to show detailed errors when
xmlstarlet is the validator. It's now run once with the '-q'
option to suppress harmless/expected messages and if it actually
fails, it's run again without '-q' but with '-e' to show
the actual errors.
Change-Id: I4bdc9d2ea6741e8d2e5eb82df60c68ccc59e1f5e
Moved the xmldoc build logic from the top-level Makefile into
its own script "make_xml_documentation" in the build_tools
directory.
Created a new utility script "get_sourceable_makeopts", also in
the build_tools directory, that dumps the top-level "makeopts"
file in a format that can be "sourced" from shell sscripts.
This allows scripts to easily get the values of common make
build variables such as the location of the GREP, SED, AWK, etc.
utilities as well as the AST* and library *_LIB and *_INCLUDE
variables.
Besides moving logic out of the Makefile, some optimizations
were done like removing "third-party" from the list of
subdirectories to be searched for documentation and changing some
assignments from "=" to ":=" so they're only evaluated once.
The speed increase is noticeable.
The makeopts.in file was updated to include the paths to
REALPATH and DIRNAME. The ./conifgure script was setting them
but makeopts.in wasn't including them.
So...
With this change, you can now place documentation in any"c"
source file AND you can now place it in a separate XML file
altogether. The following are examples of valid locations:
res/res_pjsip.c
Using the existing /*** DOCUMENTATION ***/ fragment.
res/res_pjsip/pjsip_configuration.c
Using the existing /*** DOCUMENTATION ***/ fragment.
res/res_pjsip/pjsip_doc.xml
A fully-formed XML file. The "configInfo", "manager",
"managerEvent", etc. elements that would be in the "c"
file DOCUMENTATION fragment should be wrapped in proper
XML. Example for "somemodule.xml":
<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
<docs>
<configInfo>
...
</configInfo>
</docs>
It's the "appdocsxml.dtd" that tells make_xml_documentation
that this is a documentation XML file and not some other XML file.
It also allows many XML-capable editors to do formatting and
validation.
Other than the ".xml" suffix, the name of the file is not
significant.
As a start... This change also moves the documentation that was
in res_pjsip.c to 2 new XML files in res/res_pjsip:
pjsip_config.xml and pjsip_manager.xml. This cut the number of
lines in res_pjsip.c in half. :)
Change-Id: I486c16c0b5a44d7a8870008e10c941fb19b71ade
Currently, each module that uses libcurl duplicates the standard
Asterisk curl user agent.
This adds a global macro for the Asterisk user agent used for
curl requests to eliminate this duplication.
ASTERISK-29861 #close
Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d
If Subscription refresh occurred between when the batched notification
was scheduled and the serialized notification was to be sent,
then new schedule notification task would never be added.
There are 2 threads:
thread #1. ast_sip_subscription_notify is called,
if notification_batch_interval then call schedule_notification.
1.1. The schedule_notification checks notify_sched_id > -1
not true, then
send_scheduled_notify = 1
notify_sched_id =
ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb....
1.2. The sched_cb pushes task serialized_send_notify to serializer
and returns 0 which means no reschedule.
1.3. The serialized_send_notify checks send_scheduled_notify if it's false
the just returns. BUT notify_sched_id is still set, so no more ast_sched_add.
thread #2. pubsub_on_rx_refresh is called
2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer
2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout
which calls send_notify
2.3. The send_notify set send_scheduled_notify = 0;
The serialized_send_notify should always unset notify_sched_id.
ASTERISK-29904 #close
Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875
Whereas BLFs allow to show a display name for each RLS entry,
the asterisk provides only the extension now.
This is not end user friendly.
This commit adds a new resource_list option, resource_display_name,
to indicate whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.
ASTERISK-29891 #close
Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681
The configObject tag contains a default attribute which
allows the default value to be specified, if applicable.
This allows for the default value to show up specially on
the wiki in a way that is clear to users.
There are a couple places in the tree where default values
are included in the description as opposed to as attributes,
which means these can't be parsed specially for the wiki.
These are changed to use the attribute instead of being
included in the text description.
ASTERISK-29898 #close
Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755
added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO
which fixes issue with module crashes on load "FRACK!, Failed assertion"
ASTERISK-29871
Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096
In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess
that hasn't been initialized, it'll assert and abort. If
digest_create_request_with_auth() fails to find the proper
auth object however, it jumps to its cleanup which does exactly
that. So now we no longer attempt to call pjsip_auth_clt_deinit()
if we never actually initialized it.
ASTERISK-29888
Change-Id: Ib6171c25c9fe8e61cc8d11129e324c021bc30b62
Fixes some minor logic issues with the module:
Previously, the OPT_END_FILTER flag was getting
tested before options were parsed, so it could
never evaluate to true (wrong ordering).
Additionally, the initially parsed timeout (float)
needs to be compared with 0, not the result int
which is set afterwards (wrong variable).
ASTERISK-29857 #close
Change-Id: I0062bce3b391c15e5df7a714780eeaa96dd93d4c
When generating dtmfs, asterisk can incorrectly think packet loss
occured during the dtmf generation, resulting in a jump in sequence
numbers when forwarding voice frames resumes. This patch forces
asterisk to re-learn the expected sequence number after each DTMF
to avoid this
ASTERISK-29869 #close
Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853
Previously there was no way to specify a connection timeout when
attempting to connect a websocket client to a server. This patch
makes it possible to now do such.
Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d
The code currently checks to see if an RFC3389
warning flag is set, except if it is, it merely
sets the flag again, the logic of which doesn't
make any sense.
This adjusts the if comparison to check if the
flag has NOT been set, and if so, emit a notice
log event and set the flag so that future frames
do not cause an event to be logged.
ASTERISK-29856 #close
Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1
Neither pjsip_message_filter's filter_on_tx_message() nor
res_pjsip_session's session_outgoing_nat_hook() were multipart
aware and just assumed that an SDP would be the only thing in
a message body. Both were changed to use the new
pjsip_get_sdp_info() function which searches for an sdp in
both single- and multi- part message bodies.
ASTERISK-29813
Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56
Added two new functions to assist checking media types...
* ast_sip_are_media_types_equal compares two pjsip_media_types.
* ast_sip_is_media_type_in tests if one media type is in a list
of others.
Added static definitions for commonly used media types to
res_pjsip.h.
Changed several modules to use the new functions and static
definitions.
ASTERISK_29813
(not ready to close)
Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9
There are times when you need to troubleshoot issues with bundled
pjproject or add new features that need to be pushed upstream
but...
* The source directory created by extracting the pjproject tarball
is not scanned for code changes so you have to keep forcing
rebuilds.
* The source directory isn't a git repo so you can't easily create
patches, do git bisects, etc.
* Accidentally doing a make distclean will ruin your day by wiping
out the source directory, and your changes.
* etc.
This commit makes that easier.
See third-party/pjproject/README-hacking.md for the details.
ASTERISK-29824
Change-Id: Idb1251040affdab31d27cd272dda68676da9b268
The ast_rtp_codecs_payloads functions do not preserve the order in which
the payloads were specified on an incoming SDP media line. This leads to
a problem with the codec negotiation functionality, as the format
capabilities of the stream are extracted from the ast_rtp_codecs. This
commit moves the ast_rtp_codec to ast_format conversion to the place
where the order is still known.
ASTERISK-28863
ASTERISK-29320
Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25
res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined
that rtpstart was configured to be an odd value. Also adding a loop
counter to prevent a possible infinite loop when looking for a free
port.
ASTERISK-27406
Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b
Newer versions of spandsp did refactoring of code to add new features
like color FAXing. This refactoring broke backwards compatibility.
Add support for the new version while retaining support for 0.0.6.
ASTERISK-29729 #close
Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1
Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.
This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.
ASTERISK-29777 #close
Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b
Makes basic call progress tone detection available
in a tech-agnostic manner with the addition of the
ToneScan application. This can determine if the channel
has encountered a busy signal, SIT tones, dial tone,
modem, fax machine, etc. A few basic async progress
tone detect options are also added to the TONE_DETECT
function.
ASTERISK-29720 #close
Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90
Fixes four misuses of the parameter 'name'. Additionally, for
consistency and to avoid such an issue in future, those few other
places, which used '\file name', were changed just to '\file'. Then,
Doxygen uses the name of the current file.
ASTERISK-29733
Change-Id: I0c18b4c863c6988b138c77448057349a9ee7052d
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.
ASTERISK-29703 #close
Change-Id: I8f3c79032d9ea1a21d16f8e11f22bd8d887738a1
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.
Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7
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.
Change-Id: I0f7de17cb411021c09fb03988bc2b904e1380192
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.
ASTERISK-29402
Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557
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.
ASTERISK-29525
Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784
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.
ASTERISK-29668
Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a
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.
ASTERISK-29275 #close
Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f
Some code has been added referencing symbols defined in a block
protected by #ifdef HAVE_PJPROJECT. Protect those code parts in
ifdef blocks too.
ASTERISK-29660
Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f
Adds parsing of ANI II digits (Originating
Line Information) to PJSIP, on par with
what currently exists in chan_sip.
ASTERISK-29472
Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847
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.
ASTERISK-29546
Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26
With gcc 11, res/res_snmp.c and res/snmp/agent.c need the
-fPIC option added to its _ASTCFLAGS.
ASTERISK-29634
Change-Id: I34649c85e075fd954e578378fabf798c3f038f50
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.
ASTERISK-29625
Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a
This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.
The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.
ASTERISK-29508 #close
Change-Id: I7955a046293f913ba121bbd82153b04439e3465f
Allow mapping pjproject log messages to the Asterisk TRACE
log level. The defaults were also changes to log pjproject
levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6
all went to DEBUG.
ASTERISK-29582
Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d
Commit 305ce3d added -Wno-parentheses-equality to Makefile.rules,
turning the previous two warning suppressions from commit e9520db
redundant. Let us remove the latter.
Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac
app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.
ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29558
ASTERISK-29567
ASTERISK-29572
Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131
PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request.
It may be used to get all X- headers in case the actual set and names of headers unknown.
ASTERISK-29389
Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b
Meter types are not well supported,
lacking support in telegraf, datadog and the official statsd servers.
We deprecate meters and provide a compliant fallback for any existing usages.
A flag has been introduced to allow meters to fallback to counters.
ASTERISK-29513
Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7
Asterisk first looks at the end of the URL to determine the file
extension of the returned audio, which in many cases will not work
because the URL may end with a query string or a URL fragment. If that
fails, Asterisk then looks at the Content-Type header and then finally
parses the URL to get the extension.
The order has been changed such that we look at the Content-Type
header first, followed by looking for the extension of the parsed
URL. We no longer look at the end of the URL, which was error prone.
ASTERISK-29527 #close
Change-Id: I1e3f83b339ef2b80661704717c23568536511032
If a re-INVITE is received after we have sent a BYE request then it
is possible for no channel to be present on the session. If this
occurs we allow PJSIP to produce the offer instead. Since the call
is being hung up if it produces an incorrect offer it doesn't
actually matter. This also ensures that code which produces SDP
does not need to handle if a channel is not present.
ASTERISK-29381
Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042
Verify `ast_check_hangup` before looping to the next sound file.
If the call is already hangup we just break the cycle.
It also ensures that the PlaybackFinished event is sent if the call was hangup.
This is also use-full when we are playing a big list of file for a channel that is hangup.
Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played.
With the patch we just break the playback cycle when the chan is hangup.
ASTERISK-29501 #close
Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8
From RFC 8225 Section 5.2.1:
The "dest" claim is a JSON object with the claim name of "dest"
and MUST have at least one identity claim object. The "dest"
claim value is an array containing one or more identity claim JSON
objects representing the destination identities of any type
(currently "tn" or "uri"). If the "dest" claim value array
contains both "tn" and "uri" claim names, the JSON object should
list the "tn" array first and the "uri" array second. Within the
"tn" and "uri" arrays, the identity strings should be put in
lexicographical order, including the scheme-specific portion of
the URI characters.
Additionally, make it clear that there was a failure to sign the JWT
payload and not necessarily a memory allocation failure.
Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9
Use cURL's URL parsing API, falling back to the urlparser library, to
parse playback URLs in order to find their file extensions.
For backwards compatibility, we first look at the full URL, then at
any Content-Type header, and finally at just the path portion of the
URL.
ASTERISK-27871 #close
Change-Id: I16d0682f6d794be96539261b3e48f237909139cb
Add check that data parameter specified when audiosocket used for externalMedia.
ASTERISK-29514 #close
Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617
In f8b0c2c9 we added support for port numbers in 'match' statements
but neglected to include that support in the PJSIP config wizard.
The removed code would have also prevented IPv6 addresses from being
successfully used in the config wizard as well.
ASTERISK-29503 #close
Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d
When we try to play a list of sound files in the same Play command,
we get only one PlaybackFinish event, after all sounds are played.
But in the case where the Play fails (because channel is destroyed
for example), Asterisk will send one PlaybackFinish event for each
sound file still to be played. If the list is big, Asterisk is
sending many events.
This patch adds a failed state so we can understand that the play
failed. On that case we don't send the event, if we still have a
list of sounds to be played.
When we reach the last sound, we send the PlaybackFinish with
the failed state.
ASTERISK-29464 #close
Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322
Philip Prindeville
Mike Bradeen
Naveen Albert
George Joseph
Maximilian Fridrich
Holger Hans Peter Freyther
sungtae kim
Ben Ford
Joshua C. Colp
Alexei Gradinari
Michael Neuhauser
Kevin Harwell
Jose Lopes
Boris P. Korzun
Trevor Peirce
Shloime Rosenblum
Moritz Fain
Thomas Guebels
Mark Petersen
Sean Bright
Torrey Searle
Luke Escude
Florentin Mayer
Alexander Traud
Dustin Marquess
Josh Soref
Matthew Kern
Jean Aunis
Joseph Nadiv
Guido Falsi
Jasper Hafkenscheid
sungtae kim
Sebastien Duthil
Igor Goncharovsky
Rijnhard Hessel
Andre Barbosa