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
There is an option to silence voicemail instructions but it does not
take into consideration if a recorded greeting exists or not. Add a
new 'S' option that does that.
ASTERISK-29632 #close
Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4
Adds an information element for ANI2 so that
Originating Line Information can be transmitted
over IAX2 channels.
ASTERISK-29605 #close
Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2
Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.
ASTERISK-18454 #close
Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b
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
Prevents reloads of app_queue from also resetting
queue statistics.
Also preserves individual queue agent statistics
if we're just reloading members.
ASTERISK-28701
Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1
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
The Milliwatt application uses incorrect tone timings
that cause it to play the 1004 Hz tone constantly.
This adds an option to enable the correct timing
behavior, so that the Milliwatt application can
be used for milliwatt test lines. The default behavior
remains unchanged for compatability reasons, even
though it is incorrect.
ASTERISK-29575 #close
Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c
Previously, the Morsecode application only supported international
Morse code. This adds support for American Morse code and adds an
option to configure the frequency used in off intervals.
Additionally, the application checks for hangup between tones
to prevent application execution from continuing after hangup.
ASTERISK-29541
Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4
Adds a function to scramble audio on a channel using
whole spectrum frequency inversion. This can be used
as a privacy enhancement with applications like
ChanSpy or other potentially sensitive audio.
ASTERISK-29542
Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e
A list of codecs to use for dialplan-originated calls can
now be specified in Originate, similar to the ability
in call files and the manager action.
Additionally, we now default to just using the slin codec
for originated calls, rather than all the slin* codecs up
through slin192, which has been known to cause issues
and inconsistencies from AMI and call file behavior.
ASTERISK-29543
Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883
Adds function to selectively drop specified frames
in the TX or RX direction on a channel, including
control frames.
ASTERISK-29478
Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec
Allows multiple files comprising an agent announcement
to be played by separating on the ampersand, similar
to the multi-file support in other Asterisk applications.
ASTERISK-29528
Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a
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
Adds application to asynchronously collect digits
dialed on a channel in the TX or RX direction
using a framehook and stores them in a specified
variable, up to a configurable number of digits.
ASTERISK-29477
Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f
While several applications exist to wait for
a certain event to occur, none allow waiting
for any generic expression to become true.
This application allows for waiting for a condition
to become true, with configurable timeout and
checking interval.
ASTERISK-29444
Change-Id: I08adf2824b8bc63405778cf355963b5005612f41
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
Hitherto, the A option has made it possible to play
audio upon answer to the called party only. This option
is expanded to allow for playback of an audio file to
the caller instead of or in addition to the audio
played to the answerer.
ASTERISK-29442
Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e
Caller ID can now be set on the called channel and
Variables can now be set on the destination
using the Originate application, just as
they can be currently using call files
or the Manager Action.
ASTERISK-29450
Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66
Adds a new ConfKick() application, which may
be used to kick a specific channel, all channels,
or all non-admin channels from a specified
conference bridge, similar to existing CLI and
AMI commands.
ASTERISK-29446
Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b
A new user option, answer_channel, adds the capability to
prevent answering the channel if it hasn't already been
answered yet.
ASTERISK-29440
Change-Id: I26642729d0345f178c7b8045506605c8402de54b
* Implemented the new "to" parameter of the MessageSend()
dialplan application. This allows a user to specify
a complete SIP "To" header separate from the Request URI.
* Completely refactored the get_outbound_endpoint() function
to actually handle all the destination combinations that
we advertized as supporting.
* We now also accept a destination in the same format
as Dial()... PJSIP/number@endpoint
* Added lots of debugging.
ASTERISK-29404
Reported by Brian J. Murrell
Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce
Introduces three new dialplan functions, MIN and MAX,
which can be used to calculate the minimum or
maximum of up to two numbers, and ABS, an absolute
value function.
ASTERISK-29431
Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d
Up until now, the VOLUME function has been write
only, so that TX/RX values can be set but not
read afterwards. Now, previously set TX/RX values
can be read later.
ASTERISK-29439
Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f
In multidomain environments, it is desirable to create
PJSIP endpoints with the domain info in the endpoint name
in pjsip_endpoint.conf. This resulted in an error with
registrations, NOTIFY, and OPTIONS packet generation.
This commit will detect if there is an @ in the endpoint
identifier and generate the URI accordingly so NOTIFY and
OPTIONS From headers will generate correctly.
ASTERISK-28393
Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619
By default Asterisk reports the PJSIP version in a SOFTWARE attribute
of every STUN packet it sends. This may not be desired in a production
environment, and RFC5389 recommends making the use of the SOFTWARE
attribute a configurable option:
https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2
This patch adds a `stun_software_attribute` yes/no option to make it
possible to omit the SOFTWARE attribute from STUN packets.
ASTERISK-29434
Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b
RFC 4235 Section 4.1.6 describes XML elements that should be
sent to subscribed endpoints to identify the local and remote
participants in the dialog.
This patch adds this functionality to PJSIP by iterating through the
ringing channels causing the NOTIFY, and inserts the channel info
into the dialog so that information is properly passed to the endpoint
in dialog-info+xml.
ASTERISK-24601
Patch submitted: Joshua Elson
Modified by: Joseph Nadiv and Sean Bright
Tested by: Joseph Nadiv
Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b
Hitherto, VoiceMail() played a non-customizable beep tone to indicate
the caller could leave a message. In some cases, the beep may not
be desired, or a different tone may be desired.
To increase flexibility, a new option allows customization of the tone.
If the t option is specified, the default beep will be overridden.
Supplying an argument will cause it to use the specified file for the tone,
and omitting it will cause it to skip the beep altogether. If the option
is not used, the default behavior persists.
ASTERISK-29349
Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280
Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.
This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.
ASTERISK-29380
Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81
Enhancements:
* The MessageSend dialplan application now takes an optional
third argument that can set the message's "To" field on
outgoing messages. It's an alternative to using the
MESSAGE(to) dialplan function.
NOTE: No channel driver currently implements this field. A
follow-on commit for res_pjsip_messaging will implement it for
the chan_pjsip channel driver.
* To prevent confusion with the first argument, currently named
"to", it's been renamed to "destination". Its function,
creating the request URI, hasn't changed.
* The documentation for MessageSend was updated to be
more clear about the parameters and how they interact
the MESSAGE() dialplan function.
* With the rename of MessageSend's first parameter, and the fact
that message.c references <info> elements in chan_sip.c,
res_pjsip_messaging.c and res_xmpp, they each needed
documentation updates to use MessageDestinationInfo instead of
MessageToInfo.
* appdocsxml.dtd was updated to include a missing element
declaration for "dataType". This was showing up as an error
in Eclipse's dtd editor.
* Despite the changes in this commit, there should be
no impact to current users of MessageSend.
Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a
The 'core' console (ie: asterisk -c) does read logger.conf and does
use the dateformat= option.
Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
and uses a hard coded dateformat option for printing received verbose messages:
main/logger.c: static char dateformat[256] = "%b %e %T"
This change will load logger.conf for each remote console session and
use the dateformat= option to set the per-line timestamp for verbose messages
Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
ASTERISK-25358: #close
Reported-by: Igor Liferenko
Some configuration items for a transport do not result in
the underlying transport changing, but instead are just
state we keep ourselves and use. It is perfectly reasonable
to change these items.
These include local_net and external_* information.
ASTERISK-29354
Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a
minargs enables enforcing of minimum count of arguments to pass to
func_odbc, so if you're unconditionally using ARG1 through ARG4 then
this should be set to 4. func_odbc will generate an error in this case,
so for example
[FOO]
minargs = 4
and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
potentially leaked ARG4 from Gosub().
ARGC is needed if you're using optional argument, to verify whether or
not an argument has been passed, else it's possible to use a leaked ARGn
from Gosub (app_stack). So now you can safely do
${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
Change-Id: I6ca0b137d90b03f6aa9c496991f6cbf1518f6c24
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext. You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.
* Added logic to iax2_call() and authenticate_reply() to print
a warning and hanhup the call if encryption is requested and
there's no secret or auth method. This prevents the crash.
* Added the ability to specify a default "auth" in the [general]
section of iax.conf.
ASTERISK-29624
Reported by: N A
Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254
When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received
This allows applications to perform actions based on the failure
reason.
ASTERISK-29252 #close
Reported-by: Dan Cropp
Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4
As described in the issue, /tmp is not a suitable location for a
large amount of cached media files, since most distributions make
/tmp a RAM-based tmpfs mount with limited capacity.
I opted for a location that can be configured separately, as opposed
to using a subdirectory of spooldir, given the different storage
profile (transient files vs files that might stay there indefinitely).
This commit just makes the cache directory configurable, but leaves
it at /tmp by default, to ensure backwards compatibility.
A future commit that only targets master could change the default
location to something more sensible such as /var/tmp/asterisk. At
that point, the cachedir could be created and cleaned up during
uninstall by the Makefile script.
ASTERISK-29143
Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d
When using this option, answering the channel is deferred until
all prompts/greetings have been played and the caller is about
to leave their message.
ASTERISK-29118 #close
Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb
