Telephone system:autopatch

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Project: Telephone system:autopatch
Featured: No
State Completed
Members xopr
GitHub No GitHub project defined. Add your project here.
Description use a VoIP system to communicate over radio
Picture
Autopatch.jpg

To connect the PABX to an external communications channel, one needs a tiny bit of hardware and some scripting to get it working. A proof of concept is made that used the PMRs in the space and a couple of components on a perf-board.


synopsis

Inspired by freeswitch.org's Examples Ham Radio, I wanted to be able to communicate over the air using FreeSWITCH.

It's a simple script with the least amount of hardware you can think of.

implementation

hardware

Using Stabo freecomm 450 portable transceivers
FreeSWITCH autopatch schematic

perl script

#!/usr/bin/perl

use strict;
use Switch;
use Data::Dumper;
use POSIX qw(strftime);
use Device::SerialPins;

my $conference_name = 'radio';
my $roger_beep      = 'tone_stream://%(60,0,2000);%(60,0,1200);%(60,0,800);% (80,0,1500)';
my $serial_port     = '/dev/ttyUSB0';

our $session;

my $ptt_port        = new Device::SerialPins( $serial_port );
my $api             = new freeswitch::API;
my $con             = new freeswitch::EventConsumer( 'CUSTOM' ); #all
my $nTalkers        = 0;

resetTransmission();

freeswitch::consoleLog( 'INFO', "Registered on CUSTOM events\n" );

while( $session->ready )
{
  # Check every half a second to prevent a small cpu from clipping
  #sleep( 0.5 ); # eats cpu
  $session->sleep( 500 );

  my $event = $con->pop;

  if ( $event
   && ($event->getHeader('Event-Name') eq 'CUSTOM' )
   && ( $event->getHeader('Event-Subclass') eq 'conference::maintenance' ))
  {
    my $action = $event->getHeader('Action');

    if ( $event->getHeader('Caller-Channel-Name') =~ m/^portaudio/ )
    {
      # portaudio event, check DTMF
      if( $action eq 'dtmf' )
      {
        my $dtmf_key = $event->getHeader('dtmf-key');
        freeswitch::consoleLog('INFO', 'DTMF: ' . $dtmf_key );
        switch( $dtmf_key )
        {
        }
      }
    }
    else
    {
      # event from other member
      switch ( $action )
      {
        case 'start-talking'
        {
          PTT( 1 );
        }
        case 'stop-talking'
        {
          PTT( 0 );
        }
        case m/^(del|mute|deaf)-member$/
        {
          # recount the number of talkers since the person throwing the horn
          # onhook might trigger a talk event before leaving
          handleTalkerCount();
        }
      }
    }
  }

  # if we have 2 members left (portaudio and this script), hup all members and stop the conference
}

sub PTT
{
  if ( shift )
  {
    # New talker
    $nTalkers++;
    if ( $nTalkers == 1 )
    {
      freeswitch::consoleLog('INFO', "Start of transmission\n" );
      $ptt_port->set_rts( 1 );
    } 
  } else {
    # Talker done
    $nTalkers--;

    if ( $nTalkers == 0 )
    {
      resetTransmission();
    }
  }

  # resetTransmission could change the number of talkers
  if ( $nTalkers < 0 )
  {
    freeswitch::consoleLog('WARNING', "Number of talkers was below zero, resetting..\n" );
    $nTalkers = 0;
  }
}

sub resetTransmission
{
  # reset the number of talkers
  $nTalkers = 0;

  # roger beep
  $api->execute( 'conference', $conference_name.' play '.$roger_beep );
  $session->sleep( 500 );
  #sleep( 0.7 ); # eats cpu

  # end of transmit
  $ptt_port->set_rts( 0 );

  freeswitch::consoleLog('INFO', "End of transmission\n" );
}

sub handleTalkerCount
{
  freeswitch::consoleLog('INFO', "Member event, counting number of talkers..\n" );

  my $conference_info = $api->execute( 'conference', $conference_name.' list' );
  my $nRealTalkers =()= $conference_info =~ /[;\|]talking[;\|]/gi;
  my $nMembers = ($conference_info =~ tr/\n//);

  # adjust number of talkers
  if ( $nRealTalkers < $nTalkers )
  {
    $nTalkers = $nRealTalkers + 1;
    PTT( 0 );
  }

  if ( $nMembers <= 2 )
  {
    freeswitch::consoleLog('INFO', "No humans in the conference anymore, terminating script..\n" );
    $session->hangup();
  }
}

1; # Return value


dial plan

The dialplan consists of two extensions: a conference where phones can dial into (1310), and an extension which runs the perl script. The conference requires a minimum of two participants. When the first person calls, the conference will automatically call the script extension and portaudio device which will be bridged directly (auto answer).

<extension name="test porto">
  <condition field="destination_number" expression="^1310$">
    <action application="answer"/>
    <action application="conference_set_auto_outcall" data="portaudio/auto_answer"/>
    <action application="conference_set_auto_outcall" data="loopback/1311"/>
    <action application="conference" data="radio@ultrawideband+flags{mintwo}"/>
  </condition>
</extension>

<extension name="test porto event">
  <condition field="destination_number" expression="^1311$">
    <action application="answer"/>
    <action application="perl" data="autopatch.pl"/>
  </condition>
</extension>