Adds an option that will play an audio file
to the party while AMD is running on the
channel, so the called party does not just
hear silence.
ASTERISK-30179 #close
Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6
Adds the EXPORT function, which allows write
access to variables and functions on other
channels.
ASTERISK-29432 #close
Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf
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
Adds TRIM, LTRIM, and RTRIM, which can be used
for trimming leading and trailing whitespace
from strings.
ASTERISK-30222 #close
Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554
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
This change adds an option, answeredonly, that will prevent music on
hold on channels that are not answered.
ASTERISK-30135
Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08
Adds an AMI event to indicate that a deadlock
has likely started, when Asterisk is compiled
with DETECT_DEADLOCKS enabled. This can make
it easier to perform automated deadlock detection
and take appropriate action (such as doing a core
dump). Unlike the deadlock warnings, the AMI event
is emitted only once per deadlock.
ASTERISK-30161 #close
Change-Id: Ifc6ed3e390f8b4cff7f8077a50e4d7a5b54e42fb
Adds the end_marked_any option, which can be used
to kick a user from a conference if any marked user
leaves.
ASTERISK-30211 #close
Change-Id: I9e8da7ccb892e522546c0f2b5476d172e022c2f5
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
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
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
* 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
Not sure how but the res_geolocation.txt and
res_pjsip_geolocation.txt files seem to be orphaned and need
to be removed. This is a 16-only commit.
ASTERISK_30167 #noclose
Change-Id: I7718ffc071a91918af7558a983ccaaf257b77238
Adds the DBGetTree action, which can be used to
retrieve all of the DB keys beginning with a
particular prefix, similar to the capability
provided by the database show CLI command.
ASTERISK-30136 #close
Change-Id: I3be9425e53be71f24303fdd4d2923c14e84337e6
Adds a POLARITY function which can be used to
retrieve the current polarity of an FXS channel
as well as set the polarity of an FXS channel
to idle or reverse at any point during a call.
ASTERISK-30000 #close
Change-Id: If6f50998f723e4484bf68e2473f5cedfeaf9b8f1
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
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
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
Adds an option to VoiceMailMain that prevents the user
from deleting messages during that application invocation.
This can be useful for public or shared mailboxes, where
some users should be able to listen to messages but not
delete them.
ASTERISK-30063 #close
Change-Id: Icdfb8423ae8d1fce65a056b603eb84a672e80a26
An m option to Park and ParkAndAnnounce now allows
specifying a music on hold class override.
ASTERISK-30087
Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e
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
Adds the CONFBRIDGE_CHANNELS function which can be used
to retrieve a comma-separated list of channels, filtered
by a particular type of participant category. This output
can then be used with functions like UNSHIFT, SHIFT, POP,
etc.
ASTERISK-30036 #close
Change-Id: I1950aff932437476dc1abab6f47fb4ac90520b83
Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio
file.
ASTERISK-29931
Added by Michael Cargile
Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2
Currently, if any custom ring cadences are specified, they are
appended to the array of cadences from wherever we left off
last time. This works properly the first time, but on subsequent
dahdi restarts, it means that the existing cadences are left
alone and (most likely) the same cadences are then re-added
afterwards. In short order, the cadence array gets maxed out
and the user begins seeing warnings that the array is full
and no more cadences may be added.
This buggy behavior persists until Asterisk is completely
restarted; however, if and when dahdi restart is run again,
then the same problem is reintroduced.
This fixes this behavior so that cadence parsing is more
idempotent, that is so running dahdi restart multiple times
starts adding cadences from the beginning, rather than from
wherever the last cadence was added.
As before, it is still not possible to revert to the default
cadences by simply removing all cadences in this manner, nor
is it possible to delete existing cadences. However, this
does make it possible to update existing cadences, which
was not possible before, and also ensures that the cadences
remain unchanged if the config remains unchanged.
ASTERISK-29990 #close
Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d
Adds the DB_KEYCOUNT function, which can be used to retrieve
the number of keys at a given prefix in AstDB.
ASTERISK-29968 #close
Change-Id: Ib2393b77b7e962dbaae6192f8576bc3f6ba92d09
Some command line options to Asterisk only apply when Asterisk
is started and cannot be used with remote console mode. If a
user tries to use any of these, they are currently simply
silently ignored.
This prints out a warning if incompatible options are used,
informing users that an option used cannot be used with remote
console mode. Additionally, some clarifications are added to
the help text and man page.
ASTERISK-22246
ASTERISK-26582
Change-Id: I980a5380ef2c19e8ea348596396d5382893c4337
This adds the EVAL_EXTEN function, which may be used to retrieve
the variable-substituted data at any extension.
ASTERISK-29486
Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432
If Asterisk receives a SIP REFER with Session-Timers UAC
maintain Session-Timers when sending UPDATE"
ASTERISK-29843
Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c
PJSIP currently is capable of receiving flash events
and converting them to FLASH control frames, but it
currently lacks support for doing the reverse: taking
a FLASH control frame and converting it into a flash
event in the SIP domain.
This adds the ability for PJSIP to process flash control
frames by converting them into the appropriate SIP INFO
message, which can then be sent to the peer. This allows,
for example, flash events to be sent between Asterisk
systems using PJSIP.
ASTERISK-29941 #close
Change-Id: I1590221a4d238597f79672fa5825dd4a920c94dd
Adds the dialplan eval function commands to evaluate a dialplan
function from the CLI. The return value and function result are
printed out and can be used for testing or debugging.
ASTERISK-29820 #close
Change-Id: I833e97ea54c49336aca145330a2adeebfad05209
added new global config option "allow_sending_180_after_183"
that if enabled will preserve 180 after a 183
ASTERISK-29842
Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18
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
Some databases depending on their configuration using backslashes
for escaping. When combined with the use of ' this can result in
a broken func_odbc query.
This change adds a SQL_ESC_BACKSLASHES dialplan function which can
be used to escape the backslashes.
This is done as a dialplan function instead of being always done
as some databases do not require this, and always doing it would
result in incorrect data being put into the database.
ASTERISK-29838
Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d
Adds the m option to the Queue application, which allows a
music on hold class to be specified at runtime which will
override the class configured in queues.conf.
This option functions like the m option to Dial.
ASTERISK-29876 #close
Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7
Adds an option to the ReceiveMF application to allow specifying a
maximum number of digits.
Originally, this capability was not added to ReceiveMF as it was
with ReceiveSF because typically a ST digit is used to denote that
sending of digits is complete. However, there are certain signaling
protocols which simply transmit a digit (such as Expanded In-Band
Signaling) and for these, it's necessary to be able to read a
certain number of digits, as opposed to until receiving a ST digit.
This capability is added as an option, as opposed to as a parameter,
to remain compatible with existing usage (and not shift the
parameters).
ASTERISK-29877 #close
Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b
(cherry picked from commit 0da713168d)
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
This adds a new AMI action called QueueWithdrawCaller.
This AMI action makes it possible to withdraw a caller from a queue,
in a safe and a generic manner.
This can be useful for retrieving a specific call and
dispatching it to a specific extension.
It works by signaling the caller to exit the queue application
whenever it can. Therefore, it is not guaranteed
that the call will leave the queue.
ASTERISK-29909 #close
Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec
The disabledevents setting has been added to the general section
in manager.conf, which allows users to specify events that
should be globally disabled and not sent to any AMI listeners.
This allows for processing of these AMI events to end sooner and,
for frequent AMI events such as Newexten which users may not have
any need for, allows them to not be processed. Additionally, it also
cleans up core debug as previously when debug was 3 or higher,
the debug was constantly spammed by "Analyzing AMI event" messages
along with a complete dump of the event contents (often for Newexten).
ASTERISK-29853 #close
Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205
