AudioSocket

Since version BrandMeister Core 20211013-125527 has support of Asterisk's AudioSocket.

AudioSocket implementation on the Core side makes and receives DMR calls. Outgoing calls are based on VAD activity, call parameters should be predefined. On the Asterisk side all seems like a single duplex outgoing call. You are able to create more complex logic such DMR to phone and back by scripting capabilities on the both sides.

To use AudioSockets you have to configure each channel profile in Core's config. UUID is an optional parameter. When undefined, Core will generate new one on the start (each AudioSocket channel correspond to Core's connection context). Core allows only one connection per channel. New connection to the same channel will replace existing one.

To secure the system, Core authorises each connection by IP and UUID. Address can be IP or domain name. Core updates it only on start. You are able to update it in runtime via D-BUS call invokeCommand(). Also configuration of allowed DMR IDs and call type can be changed in runtime by call setSpecificValue(). See information bellow.

You can generate static UUIDs here - https://www.uuidgenerator.net

We strongly recommend you to use it on top of phone conference bridges with automatic dialout to BrandMeister Core

If you want to make a kind of telephony patch application that should initiate phone call by DMR, you can create a Registry plugin which has to initiate a call on the Asterisk side my using ARI (https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI)

Asterisk simple dialplan example

exten = 101,1,Verbose("Call to AudioSocket via Channel interface")
same = n,Answer()
same = n,Dial(AudioSocket/server.example.com:9092/6c7a28ca-4d20-4db3-9a8a-497594de57a8)
same = n,Hangup()

BrandMeister Core configuration example

AudioSocket :
{
  port = 9092;  // TCP port
  channels =
  [
    "Socket20"
  ];
};

Socket20 :
{
  // AudioSocket ID
  number = 20;
  // AudioSocket UUID (optional)
  identifier = "6c7a28ca-4d20-4db3-9a8a-497594de57a8";
  // Codec type:
  // AMBEServer - to use with AMBEserver or dvemu (please check https://github.com/dl5di/OpenDV/tree/master/DummyRepeater)
  // USB Dongle - to use with DVSI USB-3000/USB-3003/USB-3012, NW ThumbDV/ThumbDV-3 or DV3K Dongle
  type = "AMBEServer";
  // Location of codec:
  // <domain name>[:<port>][,<interval>] - in case of AMBEServer (interval - address refresh interval in seconds, 10 minutes by default)
  // <path to TTY device>[;speed=230400] - in case of USB Dongle
  location = "localhost:2460";
  // Address of Asterisk server (used to authorise incoming connection)
  address = "localhost";
  // VAD parameters
  start = 47;    // Level percentage to start
  release = 10;  // Level percentage to continue
  // Outgoing session parameters (optional)
  mode = "Group";
  source = 1;
  destination = 9504;
};

API commands

Please look at D-BUS API

invokeCommand(context, command)

set alias <talker alias>
set address <asterisk address, domain names accepted>

setSpecificValue(context, parameter, value) / getSpecificValue(context, parameter)

#define VALUE_SOCKET_OUTGOING_MODE       0 
#define VALUE_SOCKET_OUTGOING_SOURCE_ID  1
#define VALUE_SOCKET_OUTGOING_TARGET_ID  2
#define VALUE_SOCKET_INCOMING_MODE       3
#define VALUE_SOCKET_INCOMING_SOURCE_ID  4
#define VALUE_SOCKET_INCOMING_TARGET_ID  5
#define VALUE_SOCKET_HOOK_MODE           6
#define VALUE_SOCKET_CONNECTION_STATE    7

#define SOCKET_MODE_PRIVATE  0 
#define SOCKET_MODE_GROUP    1

#define SOCKET_HOOK_NONE       0
#define SOCKET_HOOK_CALL_BACK  1
#define SOCKET_HOOK_TARGET_ID  2

Note about hook mode

SOCKET_HOOK_TARGET_ID - Core uses pre-defined outgoing caller ID for private call or group ID for group call to hook calls automatically
SOCKET_HOOK_CALL_BACK - the same as SOCKET_HOOK_TARGET_ID, but also checks source call ID for private calls (usable in case of private call bridging, locks routing to the pair of source ID + destination ID)
SOCKET_HOOK_NONE - don't hook automatically, calls can be routed by Lua or Registry

HTTP API for Asterisk

HTTP API is another way to change/read DMR call settings from Asterisk dialplan. API can be called with the same prerequisites as a AudioSocket.

Call semantic

GET http(s)://<core's address>:<core's http(s) port>/socket/<audiosocket uuid>[?<url-encoded query parameters>]

Query parameters:

source - set source ID for outgoing call and hook
destination - set destination ID for outgoing call and hook
mode - set outgoing call mode
hook - change hook mode (should be a number)
alias - set talker alias for outgoing call

All query parameters are optional, BrandMeister will modify corresponding values only for existing parameters. The result of call is a JSON structure with current values of parameters. Check API section above for parameter meanings and values.

Dialplan example

exten = 102,1,Answer()
same = n,Set(CURL_RESULT=${CURL(http://server.example.com:8080/socket/6c7a28ca-4d20-4db3-9a8a-497594de57a8?source=${CALLERID(num)}&alias=${URIENCODE(${CALLERID(name)})})})
same = n,Dial(AudioSocket/server.example.com:9092/6c7a28ca-4d20-4db3-9a8a-497594de57a8)
same = n,Hangup()

exten = 103,1,Answer()
same = n,Set(CURL_RESULT=${CURL(http://server.example.com:8080/socket/6c7a28ca-4d20-4db3-9a8a-497594de57a8)})
same = n,Set(source_id=${JSONELEMENT(CURL_RESULT,incoming/source)})
same = n,Verbose(source_id)

Returned JSON

{
  "incoming" :
  {
    "mode" : 0,
    "source" : 456,
    "destination" : 123
  },
  "outgoing" :
  {
    "mode" : 0,
    "source" : 123,
    "destination" : 456
  },
  "hook" : 0
}

incoming - parameters of current or last received DMR call
outgoing - parameters in use for outgoing DMR call and hook