Llamar multiples numeros desde una base de dato

#!/bin/bash

mysql --silent -h localhost  -u root -p198dd -D asterisk<<<'select dst  from cdr where dst="100" limit 0,5' > tmp_results

while read dst
do

 echo $dst
`asterisk -x "originate SIP/$dst extension 0@internal" `


done < tmp_results


root@asterisk-dominicana:~# ./db.sh

Asterisk random caller id and rand function

Example1

 exten => s,1,Set(junky=${RAND(1,8)});
     - Sets junky to a random number between 1 and 8, inclusive.

Example2

if we had few caller ID number want to use it, try this:

[outbound]
exten => _886X.,1,Noop
  same => n,Gosub(pickCallerIDnum,cell${RAND(1,5)},1)
  same => n,Dial(SIP/${EXTEN}@gateway,32,gCX)

[pickCallerIDnum]
exten => cell1,1,Set(CALLERID(num)=09xxxxxxx1)
  same => n,Return
exten => cell2,1,Set(CALLERID(num)=09xxxxxxx2)
  same => n,Return
exten => cell3,1,Set(CALLERID(num)=09xxxxxxx3)
  same => n,Return
exten => cell4,1,Set(CALLERID(num)=09xxxxxxx4)
  same => n,Return
exten => cell5,1,Set(CALLERID(num)=09xxxxxxx5)
  same => n,Return

Asterisk perment calling

#!/bin/bash

#$1=number of calls -1

#2 pause

#3 destination  number
COUNTER=0
         while [  $COUNTER -lt $1 ]; do
             echo The counter is $COUNTER
sleep  $2


asterisk -x "originate SIP/perment/9990000$3 extension 0@internal"

let COUNTER=COUNTER+1
         done


root@asterisk-dominicana:~# ./while.sh 100 3 14795824808




dial plan
[internal]
exten=>0,1(music),Playback(/var/lib/asterisk/moh/macroform-cold_day)
same=>n,Goto(music)


 --------------------------------------------------------------
 Otras variantes mandandolo a un local channel para  poder  poner el caller ID



#!/bin/bash

#$1=number of calls -1

#2 pause

#3 destination  number
COUNTER=0
         while [  $COUNTER -lt $1 ]; do
             echo The counter is $COUNTER
sleep  $2

asterisk -x "originate Local/$3@perment extension 1701@radio"


let COUNTER=COUNTER+1
         done




########################

EJEMPLO DEL ARHICOV calleridlist.conf  del cual el script tomara los caller id
/root/calleridlist.conf
18007142585
18097142358
13052365874
################

DIAL PLAN






[perment]
;exten =>_x.,1,Gosub(pickCallerIDnum,cell${RAND(1,5)},1) ; ya no usare el gobsub para el caller id
exten=>_x.,1,Set(callid=${SHELL(shuf -n 1 /root/calleridlist.conf)}) ;tomar un caller id random de un archv
same=>n,Set(CALLERID(num)=${callid})
same=>n,Dial(SIP/99900${EXTEN}@perment,25)
same=>n,Hangup()
[pickCallerIDnum]
exten =>cell1,1,Set(CALLERID(num)=343444)
same => n,Return

exten => cell2,1,Set(CALLERID(num)=044552)
  same => n,Return

exten => cell3,1,Set(CALLERID(num)=10573)
  same => n,Return

exten => cell4,1,Set(CALLERID(num)=900094)
  same => n,Return

exten => cell5,1,Set(CALLERID(num)=10995)
  same => n,Return

[radio]
exten=>1701,1,Answer
same=>n,NoCDR()
same=>n,Set(CHANNEL(MUSICCLASS)=radio)
same=>n,MusicOnHold()
same=>n,Hangup
Luego corremos en consola

root@asterisk-dominicana:~# ./while.sh 5 5 18097143489

Asterisk Backgroun Music a during a call

[internal]
exten=>0,1,Hangup()
exten=>0,2(music),Playback(/var/lib/asterisk/moh/macroform-cold_day)
same=>n,Goto(music)

[spy-exten]
exten =>_1923,1,Dial(Local/1924@spy-exten,30,G(internal^0^1))
exten=>_1924,1,ChanSpy(SIP/100,B)

Remote Party ID

Overview

In SIP, the Remote Party ID header field enables popular services as well as some regulatory and public safety requirements.

These services include the following:

• calling identity delivery

• calling identity delivery blocking

• tracing originator of call

Sistema de Grabacion de mensaje y envio de mensaje por correo.

[recording]
exten=>s,1,Verbose(recording calls from the caller ${CALLERID(num)} )
same=>n,Set(FECHA=${STRFTIME(${EPOCH},,%Y-%m-%d_%H-%M-%S)}_${CALLERID(num)})
same=>n,Playback(beep)
same=>n,MixMonitor(/var/www/${FECHA}.wav)
same=>n,Set(TIMEOUT(absolute)=90)
same=>n,Wait(90)
exten=>h,1,System(/usr/bin/mpack -s "Asterisk Dominicana ${FECHA}" /var/www/${FECHA}.wav ambiorixg12@hotmail.com,ambiorixg12@gmail.com)

Restringir una extension a solo hacer llamadas internas en Elastix

Creamos un contexto personalizado en extension_custom.conf 


[restringido]
include => ext-local


Luego agregamos la extension a dicho contexto  y listo.

VoIP Codec: Payload size

By Nik On June 16, 2011 · In Asterisk, Mobile VoIP, SIP protocol, VoIP calls quality

The size of the payload of each encoded voice packet influences two things: lag and bandwidth.
Every encoded packet that is sent incurs fixed bandwidth overheads (due to IP and other headers added to the data in the network). Thus, larger payloads incur a proportionately smaller overhead, thus reducing the nominal bandwidth utilisation. However, by using larger payloads, more audio (ie., a longer period of time) is required to construct a single packet, which in turn increases the amount of time it takes for even the beginning of the packet to reach the other end and be decoded, thus increasing the lag in the conversation. This is a typical trade-off in VoIP.
Most codecs use payload sizes rage from 10 to 40 milliseconds per packet. Default payload and bandwidth consumption by different codecs:


http://www.voip-sip.org/voip-codec-payload-bandwidth-required/

Sistema de IVR PHP & Asterisk

;IVR
[ivr]
exten=>s,1,Set(GLOBAL(LOOPCOUNT)=1)
same=>n,verbose(value of ${LOOPCOUNT})
;same=>n,verbose(value of ${LOOPCOUNT})

same=>n,Answer()
same=>n,Background(/var/lib/asterisk/sounds/custom/6)
same=>n,Waitexten(2)

exten =>i,1,Set(CALLERID(num)=${var1})
exten=>i,2,Dial(SIP/${advnumber}@comk,15)
same=>n,Hangup()

exten =>1,1,Set(CALLERID(num)=${var1})
exten=>1,2,Dial(SIP/${advnumber}@comk,15)
same>n,Hangup()

;exten=>h,1,System(/bin/echo "this is the ${advnumber} with the ${DIALSTATUS}">/home/postback.conf)
exten=>h,1,System(curl http://obamacare-guide.org/obm-api/obm-api.php -G -d"username=htgambiorix&password=a123&phone=${var1}&answered=ANSWER")
exten=>t,1,NoOp()
same=>n,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
same=>n,verbose(value of ${LOOPCOUNT})
same=>n,GotoIf($[${LOOPCOUNT} > 2]?hangup,hang,1)
same=>n,Goto(ivr,s,1)

 #################################################################

PHP

<?php

$pbx="localhost";

$pbx="localhost";

$trunk="comk";

$src=$_GET[phone];

$dest=$_GET[advnumber];


$extension=array($src); //numeros a llamar si vamos a usar extensions internas debemos  remover la variable trunk  en la linea Channel: SIP/$value@$trunk



 foreach ($extension as $value){

 $socket = fsockopen($pbx,"5038", $errno, $errstr, $timeout);
 fputs($socket, "Action: Login\r\n");
 fputs($socket, "UserName: admin\r\n");     //
 fputs($socket, "Secret: am56\r\n\r\n");  //
                                            //
              $wrets=fgets($socket,128);
              echo $wrets;
              fputs($socket, "Action: Originate\r\n" );
               fputs($socket, "Channel: SIP/$trunk/$value\r\n" );
                #fputs($socket, "Channel: SIP/100\r\n" );
                fputs($socket, "Exten: s\r\n" );
               fputs($socket, "Context: ivr\r\n" );
               fputs($socket, "Priority: 1\r\n" );
               fputs($socket, "CallerID: $dest\r\n" );
                fputs($socket, "Variable: __var1=$src\r\n" );
              fputs($socket, "Variable: __advnumber=$dest\r\n" );
              fputs($socket, "Async: yes\r\n\r\n" );
              fputs($socket, "Action: Logoff\r\n\r\n");
 sleep (1);
 $wrets=fgets($socket,128);

}
?>