Asterisk Realtime Centos 7

Asterisk realtime centos 7

yum install epel-release

Install dependencies

yum install unixODBC-devel

yum install unixODBC

yum install python-devel

yum install python-pip

yum install MySQL-python

yum install mysql-connector-odbc

yum install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel

Install LAMP
https://hostadvice.com/how-to/how-to-install-lamp-stack-on-centos-7/

Install Asterisk


sed -i 's/\(^SELINUX=\).*/\SELINUX=disabled/' /etc/sysconfig/selinux
sed -i 's/\(^SELINUX=\).*/\SELINUX=disabled/' /etc/selinux/config
reboot
MAKE SURE SELINUX IS DISABLED running the next command
sestatus

yum -y update

yum -y groupinstall core base "Development Tools"

yum install -y make wget openssl-devel ncurses-devel newt-devel libxml2-devel kernel-devel gcc gcc-c++ sqlite-devel

cd /usr/src/

wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16-current.tar.gz

tar zxvf asterisk*

cd /usr/src/asterisk*

contrib/scripts/install_prereq install

./configure --libdir=/usr/lib64 --with-pjproject-bundled --with-jansson-bundled && make menuselect && make && make install

make samples

make config

ldconfig

service asterisk start

asterisk -rvvvvvvvvvvvvv

Install Realtime 

Create the DB

# mysqladmin -u root -p create asterisk

First, install Alembic:

# pip install alembic

Then, move to the Asterisk source directory containing the Alembic scripts:

# cd contrib/ast-db-manage/

Next, edit the config.ini.sample file and change the sqlalchemy.url option, e.g.

sqlalchemy.url = mysql://root:password@localhost/asterisk

such that the URL matches the username and password required to access your database.

Then rename the config.ini.sample file to config.ini

# cp config.ini.sample config.ini

Finally, use Alembic to setup the database tables:

# alembic -c config.ini upgrade head

You'll see something similar to:

# alembic -c config.ini upgrade head
INFO  [alembic.migration] Context impl MySQLImpl.
INFO  [alembic.migration] Will assume non-transactional DDL.
INFO  [alembic.migration] Running upgrade None -> 4da0c5f79a9c, Create tables
INFO  [alembic.migration] Running upgrade 4da0c5f79a9c -> 43956d550a44, Add tables for pjsip
#

You can then connect to MySQL to see that the tables were created:

# mysql -u root -p -D asterisk

mysql> show tables;
+--------------------+
| Tables_in_asterisk |
+--------------------+
| alembic_version    |
| iaxfriends         |
| meetme             |
| musiconhold        |
| ps_aors            |
| ps_auths           |
| ps_contacts        |
| ps_domain_aliases  |
| ps_endpoint_id_ips |
| ps_endpoints       |
| sippeers           |
| voicemail          |
+--------------------+
12 rows in set (0.00 sec)
mysql> quit

/etc/odbc.ini
	[asterisk-mysql]
	Description = MySQL connection to 'asterisk' database
	Driver = MySQL
	Database = asterisk
	Server = localhost
	UserName = root
	Password = 7891
	Port = 3306
	Socket = /var/lib/mysql/mysql.sock
	/etc/odbcinst.ini
	# Example driver definitions
	# Driver from the postgresql-odbc package
	# Setup from the unixODBC package
	[PostgreSQL]
	Description = ODBC for PostgreSQL
	Driver = /usr/lib/psqlodbc.so
	Setup = /usr/lib/libodbcpsqlS.so
	Driver64 = /usr/lib64/psqlodbc.so
	Setup64 = /usr/lib64/libodbcpsqlS.so
	FileUsage = 1
	# Driver from the mysql-connector-odbc package
	# Setup from the unixODBC package
	[MySQL]
	Description = ODBC for MySQL
	Driver = /usr/lib/libmyodbc5.so
	Setup = /usr/lib/libodbcmyS.so
	Driver64 = /usr/lib64/libmyodbc5.so
	Setup64 = /usr/lib64/libodbcmyS.so
	FileUsage = 1
	/etc/asterisk/res_odbc.conf
	[asterisk-mysql]
	enabled => yes
	dsn => asterisk-mysql
	username => root
	password => 7891
	pre-connect => yes
	/etc/asterisk/cdr_adaptive_odbc.conf
	[asterisk-mysql]
	connection=asterisk-mysql
	table=cdr
	alias start => calldate
	----------------------------
	Load on asterisk
	module reload res_odbc.so Check connection asterisk_realtime*CLI> odbc show all ODBC DSN Settings -----------------   Name:   asterisk   DSN:    asterisk     Number of active connections: 1 (out of 1)     Logging: Disabled Connecting PJSIP Sorcery to the Realtime Database nano  /etc/asterisk/sorcery.conf [res_pjsip] ; Realtime PJSIP configuration wizard endpoint=realtime,ps_endpoints auth=realtime,ps_auths aor=realtime,ps_aors domain_alias=realtime,ps_domain_aliases contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Realtime Configuration /etc/asterisk/extconfig.conf ps_endpoints => odbc,asterisk ps_auths => odbc,asterisk ps_aors => odbc,asterisk ps_domain_aliases => odbc,asterisk ps_endpoint_id_ips => odbc,asterisk ps_contacts => odbc,asterisk /etc/asterisk/modules.conf preload => res_odbc.so preload => res_config_odbc.so noload => chan_sip.so  /etc/asterisk/pjsip.conf [transport-udp] type=transport protocol=udp bind=0.0.0.0 Create endpoints insert into ps_aors (id, max_contacts) values (103, 1); insert into ps_auths (id, auth_type, password, username) values (103, 'userpass', 103, 103); insert into ps_endpoints (id, transport, aors, auth, context, disallow, allow, direct_media) values (103, 'transport-udp', '103', '103', 'testing', 'all', 'g722', 'no');

                      pjsip show endpoint 103

https://gist.github.com/ambiorixg12/8cea6702d8c2b5afef31690c0f5d72e0

https://gist.github.com/ambiorixg12/79805c7d25c7ca7d6a1cc0c009002b6d

https://gist.github.com/ambiorixg12/0811ced4784d8a7e13c0567b940da993

https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime

----------------------------php code----

php /home/ambiorix/insert_device.php 2222

<?php

$link = mysqli_connect("localhost", "root", "mypass", "asterisk");

/* check connection */

if (mysqli_connect_errno()) {

    printf("Connect failed: %s\n", mysqli_connect_error());

    exit();

}

$sip_device ="$argv[1]";

$query1 = "INSERT INTO `ps_aors` (`id`, `max_contacts`) VALUES ($sip_device, 1)";

if (mysqli_query($link, $query1)) {

    printf("Query 1 executed successfully.\n");

} else {

    printf("Error executing query 1: %s\n", mysqli_error($link));

}

$query2 = "INSERT INTO `ps_auths` (`id`, `auth_type`, `password`, `username`) VALUES ($sip_device, 'userpass', $sip_device, $sip_device)";

if (mysqli_query($link, $query2)) {

    printf("Query 2 executed successfully.\n");

} else {

    printf("Error executing query 2: %s\n", mysqli_error($link));

}

$query3 = "INSERT INTO `ps_endpoints` (`id`, `transport`, `aors`, `auth`, `context`, `disallow`, `allow`, `direct_media`) 

VALUES ($sip_device, 'transport-udp', '$sip_device', '$sip_device', 'testing', 'all', 'g722', 'no')";

if (mysqli_query($link, $query3)) {

    printf("Query 3 executed successfully.\n");

} else {

    printf("Error executing query 3: %s\n", mysqli_error($link));

}

/* close connection */

mysqli_close($link);

?>

SIP Media Flow Attribute Support

Show

Home > Product Description > VoIP > SIP Media Flow Attribute Support

Overview:

The IMG supports the media flow direction attributes sendrecv, recvonly, sendonly, and inactive. These attributes, which are interpreted from the senders perspective, set the direction of the media stream to be established in the SIP protocol. The flow attributes present in the SDP portion of the SIP messaging will be used to support the call hold/resume scenario which is explained in RFC 3264 section 6.1 and still be backwards compatible with the call hold feature of RFC 2543 section B5. Up until software version 10.5.2 the IMG supported the Call Hold scenario using the "c=0.0.0.0" line  in the SDP message to put a call on hold. With the addition of the Media Flow direction Attributes a call can now be put on hold and then the call can be resumed by sending a Re-INVITE message with both the c=0.0.0.0 and one of the Flow attributes displayed above.

Call Flow Attributes:

The following attributes are supported on the IMG. These attributes are coded in the IMG and do not need to be configured. The attributes were added to support the call hold/resume feature and are supported in the Re-INVITE message only. Below is some background information on the Media Call Flow attributes added.

sendrecv - Used to establish a 2-way media stream.

recvonly - The SIP endpoint would only receive (listen mode) and not send media.

sendonly - The SIP endpoint would only send and not receive media.

inactive -  The SIP endpoint would neither send nor receive media.

Table 1 below shows the response the IMG will use when it receives a Media Flow Attribute on the incoming side.

Table 1:

Attribute in incoming Re-INVITE	Attribute in 200 OK response	IMG Connection Mode
a=sendrecv	a=sendrecv	2-Way Connection
a=sendonly	a=recvonly	Listen Mode (recv)
a=recvonly	a=sendonly	Transmit Mode (send)
c=0.0.0.0	c=0.0.0.0	Hold
a=inactive	a=inactive	Hold
c=0.0.0.0 and a=inactive	c=0.0.0.0 and a=inactive	Hold

Call Flow Information:

Call Flow: Call Hold/Resume

The Call flow diagram above explains the Call Hold/Resume feature implemented with the Call Flow Direction Attributes.

1. A 2-way call is established from GW-A to GW-B through the IMG using normal call messaging procedures.
2. GW-A wants to put the call on hold. GW-A sends a Re-INVITE message to the IMG with the a=sendonly line in the SDP. This tells the IMG that GW-A is in a send-only mode and will not receive RTP/RTCP data.
3. The IMG then sends a 200 OK response with the a=recvonly line indicating that it will only receive RTP/RTCP data and not transmit RTP/RTCP data packets to GW-A. At this point, GW-A can only transmit to IMG.
4. The IMG then sends a Re-INVITE message to GW-B with the c=0.0.0.0 and a=inactive in the SDP. By sending these two messages, the IMG effectively puts the call on hold.
5. GW-B responds with a 200 OK telling the IMG it is now on hold. There is no voice path between IMG and GW-B.
6. GW-A then wants to take the call off hold and resume the call. It sends a Re-INVITE to the IMG with the line a=sendrecv telling the IMG it is now in a send/receive mode and the call can resume.
7. The IMG responds with 200 OK message with the a=sendrecv line in its SDP.
8. The IMG then sends a Re-INVITE to GW-B with the same a=sendrecv message in its SDP. GW-A responds with 200 OK and a=sendrecv in the SDP.
9. At this point the call resumes and RTP/RTCP data begin flowing again.

Additional Information:

• Support of the Media Call Flow Attributes is limited to the Re-INVITE method only. If a gateway sends a different method such as the UPDATE method with Media Call Flow attributes set for call hold or resume,IMG will honor the UPDATE method but not put call on hold or off hold/resume.
• Absence of a Call Flow Attribute in the Re-INVITE shall default to sendrecv.
• Call Flow attributes within an INVITE message will be ignored.
• IMG by default will use c=0.0.0.0 and a=inactive in the Re-INVITE to propagate the call hold to outgoing SIP leg.
• RTP as well as RTCP will be stopped when media is put "On Hold".
• IMG honors 'recvonly' attribute in re-invite SDP, and responds with 'sendonly' in the 200 OK SDP.
• Sample SDP messages shown below:

SDP with Flow Direction at session level:

v=0

o=Dialogic_SDP 0 1 IN IP4 10.10.1.253

s=Dialogic-SIP

c=IN IP4 10.10.1.100

t=0.0

a=sendonly

m=audio 8004 RTP/AVP 0

a=rtpmap:0 PCMU/8000

a=silenceSupp:off

SDP with flow direction attribute at media level:

v=0

o=Dialogic_SDP 0 1 IN IP4 10.10.1.253

s=Dialogic-SIP

c=IN IP4 10.10.1.100

t=0.0

m=audio 8004 RTP/AVP 0

a=rtpmap:0 PCMU/8000

a=silenceSupp:off

a=sendonly

phpagi spoof

//wait a minute to give everything a chance to settle
sleep(2);


//I had it set up to check the callerid against a set of valid peeps - but i decided i was tired of that. and it made it too complex.
//get caller id. 
$cid = $agi->parse_callerid();
$cid= $cid[username];

//streamfile that says "Enter number to spoof"
$agi->stream_file('enter_spoof');
//beep then get the resulting 10 digits - set it to spoofnumber
$result = $agi->get_data('beep', 3000, 10);
$spoofnumber= $result['result'];
$agi->verbose("Spoof Number:".$spoofnumber);
//streamfile that says "enter number to call"
$agi->stream_file('call_spoof');
//beep then get the resulting 10 digits - set it to callnumber
$result = $agi->get_data('beep', 3000, 10);
$callnumber= $result['result'];
$agi->verbose("Number to call:".$callnumber);
// set caller id to the spoofnumber
$agi->set_callerid($spoofnumber);

//call the number using whatever you got
//notice i have placed the 1 before the callnumber. this is so that i can keep everything ten digits. and cuz i am lazy
$agi->exec("Dial IAX2/yourpassword@provider/1".$callnumber);


// That is it. simple.. scary and easy
?>

phpagi conference system

#!/usr/bin/php -q
<?php
set_time_limit(30);
require('/var/www/html/project1/phpagi-2.20/phpagi.php');

error_reporting(E_ALL);
$agi = new AGI();
$agi->answer();
//http://php.net/manual/en/function.date.php
$time=date('a');

$agi->stream_file("welcome");

 $cid = $agi->parse_callerid();
$agi->verbose($cid[username]);

$agi->verbose($time);
$agi->set_variable("CONFBRIDGE(user,pin)",1212);
$agi->set_variable("CONFBRIDGE(bridge,max_members)",2);

//ConfBridge(conference[,bridge_profile[,user_profile[,menu]]])

$agi->exec("ConfBridge","111");

//$agi->exec_goto('from-internal',$value,1);
?>

[outboundtest]

exten=>100,1,answer()

;same=>n,Set(CONFBRIDGE(user,pin)=1111)

;same=>n,Set(CONFBRIDGE(bridge,max_members)=2)

exten=>100,n,agi(/var/www/html/conference/conf.php)

same=>n,hangup()

Len Execif

${LEN(${CALLERID(num)})}
 ${LEN(${CALLERID(num)})}

same=>n,ExecIf($[${LEN(${CALLERID(num)})}> 2]?Set(CALLERID(num)=${CALLERID(num):1}))