GO AMI - Asterisk Originate Example

package main

import (
    "fmt"
    "log"

    "github.com/heltonmarx/goami/ami"
)

var username, secret, host string = "test", "test", "127.0.0.1:5038"

func main() {
    //flag.Parse()

    socket, err := ami.NewSocket(host)

    orig := ami.OriginateData{}
    ////////////////////////////////////////////////
    orig.Channel = "Local/1000@leg1"
    orig.Exten = "2000"
    orig.Context = "leg2"
    orig.Priority = 1
    //orig.Application = ""
    orig.Data = ""
    orig.Timeout = 10
    //orig.CallerID = "bot"
    //orig.Variable = ""
    //orig.Account = "bot"
    orig.Async = "true"
    //orig.Codecs = "alaw&ulaw&g729"
    ////////////////////////////////////////////////

    socket, err := ami.NewSocket(host)
    if err != nil {
        log.Fatalf("socket error: %v\n", err)
    }
    if _, err := ami.Connect(socket); err != nil {
        log.Fatalf("connect error: %v\n", err)
    }
    //Generate UUID for AMI requests
    uuid, _ := ami.GetUUID()
    if err := ami.Login(socket, username, secret, "Off", uuid); err != nil {
        log.Fatalf("login error: %v\n", err)
    }
    fmt.Printf("login ok!\n")

    if res, err := ami.Originate(socket, uuid, orig); err != nil {
        println("ERROR: ", err)
        println("Response: ", res)
    }

    println("Call originated Successfully")

    //time.Sleep(5 * time.Second)

    if ami.Logoff(socket, uuid) != nil {
        println("Logoff Failed")
    } else {
        println("Logoff successful")
    }
}



/// Asterisk  Dialplan
/etc/asterisk/extensions.conf
//////////////////////////////////////////////////////
[leg1]
exten =>1000,1,NoOp(leg1)
same=>n,Answer()

[leg2]
exten =>2000,1,NoOp(leg2)
same=>n,playback(tt-monkies)




// Asterisk Manager configuration
//////////////////////////////////////////
/etc/asterisk/manager.conf

[test]
secret=test
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read=system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate
write=system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate

ping with push results in file with timestamp

ping 10.230.0.26 | while read pong; do echo "$(date +%Y-%m-%d\ %H:%M:%S): $pong"; done >>ping_test.log &

Perl NetSIP REGISTER and INVITE Example

use Net::SIP;
#use strict;
#use warnings;


my $uae_sip = '10.0.0.2:5060';
my $from = '111';
my $secret = 'abcd12';

my $to = '14168404385';


# create new agent
my $ua = Net::SIP::Simple->new(
      outgoing_proxy => $uae_sip,
      registrar => $uae_sip,
      domain => $uae_sip,
      from => $from,
      auth => [ $from, $secret ],
);


if($ua->register( expires => 60 )){
    print "REGISTER sucessful. \n";
}else {
    print "REGISTER Failed";
}

#$ua->register( expires => 60 );

#print $ua->error;
#print $ua_error;
##die();
#if ($ua->error){
#    print "Registration Failed " . $ua->error;
#}else {
#    print "SIP Registration Sucessful \n";
#}

#my $rtp_done;

my $hangup = 20;
my $stopvar = 20;
my $file = '';
my $to = '0581104414';

my $peer_hangup;

my $call = $ua->invite( $to,
    init_media => $ua->rtp( 'recv_echo', $file,0 ),
    recv_bye => \$peer_hangup,
);
if($call->error ){
    # do nothing.
    print "INVITE Failed" . $call->error . "\n";
}else{

    print "INVITE sucessful \n";
}

# mainloop until other party hangs up or we hang up after

my $stopvar;
$ua->add_timer( $hangup, \$stopvar ) if $hangup;
$ua->loop( \$stopvar,\$peer_hangup );

# timeout checking block.
if ( $stopvar ) {
    $stopvar = undef;
    $call->bye( cb_final => \$stopvar );
    $ua->loop( \$stopvar );
}

Asterisk AMI Events using Perl POE

POE is a Perl framework for reactive systems, cooperative multitasking, and network applications. Read more about POE.

I'll explain, how we can use POE event loop to process the AMI Event in a asynchronous way. We can build any application in Perl that is based on AMI Events. Like Realtime Asterisk based billing, custom application to push events for web application, or any other application. With this POE Application we have all states of the calls, include some states which we don't have in the Asterisk Dialplan.

#!/usr/bin/perl -w

use strict;

use threads;

use threads::shared;

use warnings;

use Data::Dumper;

use Config::Abstract::Ini;

use POE qw(Component::Client::Asterisk::Manager);

#AMI-POE

my $host = '10.11.22.33';

my $port = '5038';

my $user = 'user';

my $pass = 'password';

my $debug = 1;

my @threadz;

POE::Component::Client::Asterisk::Manager->new(

        Alias => 'monitor',

        RemoteHost      => $host,

        RemotePort      => $port,

        Username        => $user,

        Password        => $pass,

        CallBacks       => {

                #cb_on_event => ':all', # catchall for all manager events

                cb_on_new_channel      => { 'Event' => 'Newchannel', },

                cb_on_rename =>{'Event' => 'Rename'},

                cb_on_dial             => {'Event' => 'Dial'},

                cb_on_newexten => {'Event' => 'Newexten'},

                on_Bridge      => {'Event' => 'Bridge','Bridgestate' => 'Link'},

                on_peer_status => { 'Event' => 'PeerStatus', },

                on_new_callerid        => { 'Event' => 'Newcallerid', },

                on_ring => { 'Event' => 'Newchannel', 'State' => 'Ring', },

                on_hangup      => { 'Event' => 'Hangup','Cause-txt' => 'Normal Clearing', },

                on_cdr => { 'Event' => 'Cdr', },

                on_rtcp_received => { 'Event' => 'RTCPReceived' },

        },

        inline_states => {

                #cb_on_event    => \&on_event,

                cb_on_new_channel      => \&on_new_channel,

                cb_on_rename => \&on_rename,

                cb_on_dial             => \&on_dial,

                cb_on_newexten => \&on_newexten,

                 on_Bridge      => \&on_Bridge,

                on_answer      => \&on_answer,

                on_peer_status => \&on_peer_status,

                on_new_callerid        => \&on_new_callerid,

                on_ring => \&on_ring,

                on_hangup      => \&on_hangup,

                on_cdr => \&on_cdr,

                on_rtcp_received => \&on_rtcp_received,

        },

); POE::Kernel->run();

sub on_rtcp_received(){

        my $input = $_[ARG0];

        #print Data::Dumper->Dump([$input]);

        my $packet_loss = $input->{'PacketsLost'};

        if($packet_loss >200){

                print Data::Dumper->Dump([$input]);

                # This call has lot of packet loss

        }

}

sub on_answer{

    #print "We are in ON-Answer STATE \n";

    my $input = $_[ARG0];

    print Data::Dumper->Dump([$input]);

}

sub on_rename{

    print "----------------> on_rename() \n";

        my $input = $_[ARG0];

        print Data::Dumper->Dump([$input]);

}

sub on_Bridge{

        my $input = $_[ARG0];

        #print "on Bridge CALLER ID\n";

        print Data::Dumper->Dump([$input]);

}

sub on_dial{

        my $input = $_[ARG0];

        print "on_dial\n";

        print Data::Dumper->Dump([$input]);

}

sub on_event{

        # Print all AMI Events

        my $input = $_[ARG0];

        print Data::Dumper->Dump([$input]);

        #print "on_event\n";

}

sub on_newexten{

         my $input = $_[ARG0];

         print Data::Dumper->Dump([$input]);

        my $fu_callerid = $input->{'CallerIDNum'};

        my $ext_status = $input->{'Application'};

        if($ext_status eq 'Answer'){

               print Data::Dumper->Dump([$input]);

               my $channel = $input->{'Channel'};

               my $context = $input->{'Context'};

               my $local_ext = $input->{'Extension'};

               my $priority = $input->{'Priority'};

               print "in Answer state, $channel, $context, $local_ext, $priority \n";

        }

       #print "on_newexten\n";

}

sub on_new_channel{

    my $input = $_[ARG0];

    print "on_new_channel()\n";

    print Data::Dumper->Dump([$input]);

}

sub on_peer_status{

}

sub on_new_callerid{

    print "on_new_callerid\n";

    my $input = $_[ARG0];

    my @Dial_Event=Data::Dumper->Dump([$input]);

    my $callerid = $input->{'CallerIDNum'};

    print "on_new_callerid: $callerid \n";

}

sub on_ring{

}

sub on_hangup{

         my $input = $_[ARG0];

        print Data::Dumper->Dump([$input]);

        my $fu_callerid = $input->{'ConnectedLineNum'};

        my $cause_text = $input->{'Cause-txt'};

        my $channel  = $input->{'Channel'};

        my $tmp0 = substr($channel,0,3);

        print "leg detection...$tmp0\n";

}

sub on_cdr{

}

Webrtc based calling using sipml5 and Asterisk

* This tutorial is deprecated. Asterisk compilation is seamless with pjsip-bundled option. Asterisk compilation part is deprecated one, rest of the tutorial should work.

There are few steps to make calls using webrtc client. 

1. Install prerequisites
2. Install pjproject
3. Install Asterisk 
4. Generate wss DTLS certificates for Asterisk
5. Install Apache HTTP Server
6. Generate self signed https certificates for Apache
7. Configure Asterisk
8. Configure sipml5 client

1 - Download and install prerequisites

apt-get update

apt-get upgrade

reboot

apt-get install build-essential libncurses5-dev libxml2-dev libsqlite3-dev libssl-dev uuid-dev libjansson-dev libsrtp0-dev pkg-config unzip

cd /usr/src

wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz

wget https://github.com/sipml5/sipml5/archive/master.zip

2 -  Install pjproject

cd /usr/src/
tar -xjvf pjproject-2.4.5.tar.bz2
cd pjproject-2.4.5
./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG'
make dep
make
make install
ldconfig
ldconfig -p | grep pj
PKG_CONFIG_PATH=/usr/lib64/pkgconfig/
export PKG_CONFIG_PATH

3 -  Install Asterisk

cd /usr/src/

tar -xzvf asterisk-13-current.tar.gz

./configure --prefix=/usr --libdir=/usr/lib64

make && make install && make samples                        

ln -sf /usr/lib64/libasteriskssl.so.1 /usr/lib/libasteriskssl.so.1

ln -sf /usr/lib64/libresample.so.2 /usr/lib/libresample.so.2

ln -sf /usr/lib64/libportaudio.so.2 /usr/lib/libresample.so.2

4 -  Generate wss DTLS certificates

cd /usr/src/asterisk-13.7.0/contrib/scripts/

./ast_tls_cert -C mypbx.local -O "my local pbx" -d /etc/asterisk/keys

5 -  Install Apache HTTP Server

apt-get install apache2

6 -  Generate self signed ssl certificates

openssl req -x509 -nodes -days 365 -newkey rsa:2048 -keyout /etc/apache2/ssl/apache.key -out /etc/apache2/ssl/apache.crt

sudo a2enmod ssl

vim /etc/apache2/sites-available/default-ssl.conf

Search ssl block  <VirtualHost _default_:443> and update the following fields.

      

        ServerAdmin admin@example.com

        ServerName your_domain.com

        ServerAlias www.your_domain.com

        DocumentRoot /var/www/html       

        SSLCertificateFile /etc/apache2/ssl/apache.crt

        SSLCertificateKeyFile /etc/apache2/ssl/apache.key

a2ensite default-ssl.conf

/etc/init.d/apache2 restart

7 -  Configure Asterisk

sip.conf

[99300]

deny=0.0.0.0/0.0.0.0

dtmfmode=rfc2833

canreinvite=no

host=dynamic

trustpid=yes

sendpid=no

insecure=port,invite

type=friend

nat=no

port=5060

qualify=yes

qualifyfreq=60

transport=ws,wss,udp

avpf=yes

force_avp=yes

icesupport=yes

encryption=yes

callgroup=

pickupgroup=

permit=0.0.0.0/0.0.0.0

dial=SIP/99300

secret=c602999e72c0054c7ad2f4b94559b23d

context=users

mailbox=99300@device

callerid=300 <99300>

callcounter=yes

faxdetect=no

cc_monitor_policy=generic

dtlsenable=yes

dtlsverify=fingerprint

dtlscertfile=/etc/asterisk/keys/asterisk.pem

dtlscafile=/etc/asterisk/keys/ca.crt

dtlssetup=actpass

dtlsrekey=0

disallow=all

;allow=g729

allow=alaw

allow=ulaw

;allow=ilbc

;allow=g726

;allow=opus

[100]

type=friend

callerid="100" <100>

defaultusername=100

host=dynamic

secret=100

dtmfmode=rfc2833

insecure=invite,port

canreinvite=yes

nat=force_rport,comedia

qualify=no

context=users

disallow=all

;allow=g729

allow=alaw

allow=ulaw

;allow=ilbc

extensions.conf

[users]

exten => 100,1,Dial(SIP/100)

exten => 99300,1,Dial(SIP/99300)

http.conf

enabled=yes

bindaddr=0.0.0.0

bindport=8088

tlsenable=yes          ; enable tls - default no.

tlsbindaddr=0.0.0.0:8089    ; address and port to bind to - default is bindaddr and port 8089.

tlscertfile=/etc/asterisk/keys/asterisk.pem

tlsprivatekey=/etc/asterisk/keys/asterisk.pem

8 -  Configure sipml5 client

cd /usr/src/

unzip master.zip

mv sipml5-master/ /var/www/html/

cd /var/www/html/

mv sipml5-master/ sipml5

chown www-data:www-data sipml5 -R

Open latest Chrome/Firefox and accpet certificates.

https://asterisk-ip:8089/

https://asterisk-ip/sipml5/call.htm

Click on the advanced setting

Disable Video:

Enable RTCWeb Breaker[1]: check on

WebSocket Server URL[2]: wss://server-ip:8089/ws

SIP outbound Proxy URL[3]:udp://server-ip:5060

ICE Servers[4]: [ ]

Save the advanced setting page and nevigate to call.htm page and enter fields.

Click Register button and start making/receiving calls.

This works for my LAN environment, configure it as per your environment.

Hope it will be useful in developing your calling scenarios.