How to Configure a Digium T1/E1 Card to Use E&M Signalling

	How to Configure a Digium T1/E1 Card to Use E&M Signalling Configuring a Digium T1/E1 Card (models TE12x, TE13x, TE2xx, TE4xx, TE82x) to use E&M signalling requires editing two files: /etc/dahdi/system.conf to configure DAHDI and /etc/asterisk/chan_dahdi.conf to configure Asterisk.

Llamando un listado de numeros desde un archivo de texto

#!/bin/bash
 var=$(< number_list)
for i in $var
do
 echo "calling to $i "




   echo "calling to $i " > freecalls.logs
`asterisk -x "originate Local/$i@india extension $i@india2"`
sleep 2
done


#######################
contenido de  number_list

100
102
123
109
234



dial plan esto solo timbra la llmada porque el timeout es  5 segundos


[india]
exten=>_x.,1,verbose( calling to ${EXTEN})
same=>n,Set(CALLERID(num)=447589975480)
same=>n,Dial(SIP/+${EXTEN}@cpeak,5)

same=>n, Noop(codigo de colgado ${HANGUPCAUSE})
same=>n,hangup()

[india2]
exten=>_x.,1,Hangup()

Entendiendo el SIP TRACE

UNDERSTANDING SIP TRACES

https://supportforums.cisco.com/document/113271/understanding-sip-traces

SIP traces provide key information in troubleshooting SIP Trunks, SIP endpoints and other SIP related issues. Even though these traces are in clear text, these texts can be gibberish unless you understand fully what they mean.
This document attempts to break down each component of the SIP interaction using a practical approach. We will look at various logs, the SIP messages, headers, SDP information and try to figure out what is going on in a sip voice call transaction.
In as much as I will try to define the under lying layer of the SIP messaging, this document will not go into in-depth analysis of the SIP protocol, so it is advisable to understand SIP protocol technology to be able to understand sip traces.

Using a GoIP with A2Billing for outbound calls

A2Billing, VOIP

This is a quick guide to using a GoIP GSM gateway with A2Billing as an outbound trunk. It doesn’t go through setting up rate cards and customers in A2Billing, guides for that are already available on this site.
First some general info and tips.
This guide was written using the following. The thing most likely to cause issues is a SIP ALG in your router. NAT issues are a pain to track down!

Como Instalar Vicidial/Goautodial en Centos 5.X

This is useful for hosted servers that can't or won't install a custom ISO (GOautodial CE 3.0 ISO).

1. Install minimal version of CentOS 5.X 64bit/32bit (CentOS 5.10 is latest as of this writing)

2. Install the GOautodial yum repository

# cd /etc/yum.repos.d/
# wget http://downloads2.goautodial.org/centos/5/goautodial.repo

3. Update system and install kernel development files

# yum update -y
# yum install kernel-devel glibc-devel -y

 yum install kernel*

[root@GoAutoDial ~]# ls /lib/modules
2.6.18-308.1.1.el5  2.6.18-371.12.1.el5  2.6.18-404.el5  2.6.18-404.el5debug  2.6.18-404.el5xen


Cambiar el kernel en el panel grafico al   2.6.18-404.el5

Luego apagamos init  0 

Luego lo encendemos por el mismo panel y seguimos instalando 

Como cambiar las claves de defualt de Goautodial

Stop mysql

[root@Goautodial ~]# service mysqld stop
 

 

Step # 2: Start to MySQL server w/o password:

# mysqld_safe --skip-grant-tables 

Output:

[1] 5988
Starting mysqld daemon with databases from /var/lib/mysql
mysqld_safe[6025]: started

If  the above command get frozen  press control  + z and then reconect to mysql with no password

root@Goautodial ~]# mysql

mysql> use asterisk;

mysql> update vicidial_users set pass=8123456 where user="admin";


mysql> update phones set pass="hola" where login like '8%';

mysql> update phones set conf_secret="holar" where login like "8%";

asterisk -x " core restart now"

for restart mysql again with  password restart the mysql service 

service mysqld start

Instalando el FOP2 en FreePBX 12 Centos 6.5

 Esto fue probando en Centos 6.5 64 Bit FreePBX 12 Asterisk13 y FOP2 2.8
  108 cd /usr/src
  109  wget http://www.fop2.com/download/centos64 -O fop2.tgz
  110  tar zxvf fop2.tgz
  111  cd fop2
  112  make install
  113  /usr/local/fop2/generate_override_contexts.pl -write
  114  /usr/local/fop2/fop2_server --test
  115  ls /var/www
  116  ls /var/www/html
  117  nano /usr/local/fop2/fop2.cfg
  118  nano etc/asterisk/manager.conf
  119  nano /etc/asterisk/manager.conf
  120  nano /usr/local/fop2/fop2.cfg
  121  asterisk -rvvvvvvvvvv
  122  amportal restart
  123  asterisk -rvvvvvvvvvv
  124  service httpd restart
  125  service fop2 start
  126  service fop2 status


///////////////////////////////////
[Fri Mar 27 02:15:23.313795 2015] [core:alert] [pid 1167] [client

190.166.113.118:58108] /var/www/html/fop2/.htaccess: Invalid command 'Header', perhaps

misspelled or defined by a module not included in the server configuration
^C




ln -s /etc/apache2/mods-available/headers.load /etc/apache2/mods-enabled/

http://stackoverflow.com/questions/10185717/internal-server-error-htaccess


http://download2.fop2.com/fop2-2.29.00-debian-x86_64.tgz

chmod  -R 777 /var/www

cp  -R fop2 /var/www/


Luego vamos al  admin  
http://my-ip.com/fop2/admin
creamos una platilla  le asigamos los permisos  y luego un usuario y lo agregamos a la plantilla

Luego verificamos http://my-ip.com/fop2



Para agregar botones  nuevos  si no aparecen

[root@pbx1 fop2]# /usr/bin/php -f /var/www/html/fop2/admin/update_conf.php
Updating buttons for context GENERAL
Adding button SIP/101 into context GENERAL


[root@pbx1 fop2]# service fop2 reload
Reloading Flash Operator Panel 2 configuration:            [  OK  ]


Luego nos desloguiamos y  volvemos a entrar

ELASTIX

Avoiding Conflicts with FOP(1)

If you have the regular FOP installed and running, you will need to change the configuration for it a little bit, or disable it altogether if you do not plan to use it anymore.

To change FOP(1) configuration so you can have both versions running, edit the file op_server.cfg file that is installed in /var/www/html/panel in a regular FreePBX installation, and change the line that says:

;listen_port=4445

to:

listen_port=4444

To disable FOP(1) completely, you can edit /etc/amportal.conf and set FOPRUN=false. Then stop your currently running FOP process with the command:

FOP2 LICENSE
/usr/local/fop2/fop2_server --register --code 7uG11@y --name Asterisk-DOM



service fop2 reload

http://www.fop2.com/docs/installation.php#Installingfromtarball
http://www.fop2.com/docs/installation.php#ConfigurationServer
http://www.powerpbx.org/content/integrate-flash-operator-panel-v2-into-freepbx

Installing FreePBX 12 on CentOS 6.5

 

Initial System Setup

Icon

You MUST run all of these commands as the root user!

Icon

You MUST disable selinux.  selinux can cause strange behavior during the install

Disable selinux

In /etc/sysconfig/selinux , change the following lines:

sed -i 's/\(^SELINUX=\).*/\SELINUX=disabled/' /etc/sysconfig/selinux

reboot, and verify the selinux status by running 'sestatus'. It should say:

SELinux status: disabled

Robot Call

#!/bin/bash

mysql --silent -h localhost  -u root -p1234 -D asterisk<<<'select telephone  from dialer  where status!="called" limit 0,5' > tmp_results

while read  telephone
do

 echo $telephone
`asterisk -x "originate Local/$telephone@robot extension $telephone@voice"` &



done < tmp_results
####################

CDR Variables

1. Asterisk Project
2. …
3. Call Detail Records (CDR)

• Created by Malcolm Davenport, last modified on Aug 31, 2010

Go to start of metadata

If the channel has a CDR, that CDR has its own set of variables which can be accessed just like channel variables. The following builtin variables are available.

• ${CDR(clid)} Caller ID
• ${CDR(src)} Source
• ${CDR(dst)} Destination
• ${CDR(dcontext)} Destination context
• ${CDR(channel)} Channel name
• ${CDR(dstchannel)} Destination channel
• ${CDR(lastapp)} Last app executed
• ${CDR(lastdata)} Last app's arguments
• ${CDR(start)} Time the call started.
• ${CDR(answer)} Time the call was answered.
• ${CDR(end)} Time the call ended.
• ${CDR(duration)} Duration of the call.
• ${CDR(billsec)} Duration of the call once it was answered.
• ${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
• ${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
• ${CDR(accountcode)} The channel's account code.
• ${CDR(uniqueid)} The channel's unique id.
• ${CDR(userfield)} The channels uses specified field.

In addition, you can set your own extra variables by using Set(CDR(name)=value). These variables can be output into a text-format CDR by using the cdr_custom CDR driver; see the cdr_custom.conf.sample file in the configs directory for an example of how to do this.

Ejecutando un comando de Linux y guardar el varlo en una variable con la funcion shell

exten => 4002,1,Answer
exten => 4002,n,Set(RESULT=${SHELL(date +%s)})
exten => 4002,n,NoOp(${RESULT})
exten => 4002,n,Hangup

http://stackoverflow.com/questions/1414 ... k-dialplan

asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory

by Dan Journo on Jun.02, 2014, under Asterisk

Issue
You install Asterisk 12 successfully but receive the following error when trying to launch asterisk:
asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory
Solution
Execute this
echo “/usr/local/lib” > /etc/ld.so.conf.d/usr_local.conf
and
/sbin/ldconfig

http://www.danjourno.com/asterisk-libjansson-cannot-open-shared/

Instalando Monast en Ubuntu Server

it@ubuntu-test ~ $ python --version
Python 2.6.6

it@ubuntu-test ~ $ sudo apt-get update && sudo apt-get install python

it@ubuntu-test ~ $ sudo apt-get install apache2 php5 php-pear mc

Sistema simple de Disa y call foward

[paul]

exten=>_13052365874,1,Set(CALLERID(num)=${CALLERID(num)})
exten=>_13052365874,2,Dial(SIP/paul/18097143489,25,o)
same=>n,Hangup()

exten=>_12365478547,1,Answer()
same=>n,Authenticate(1212)
same=>n,Disa(no-password,disa)

[disa]
exten=>_x.,1,Dial(SIP/${EXTEN}@paul,25)
same=>n,hangup()