Soporte & Consultoria

Soporte Remoto y Consultoria skype : ambiorixg12.
Nota no se brinda ningun tipo de consulta o soporte fuera del blog de forma gratuita

viernes, 29 de noviembre de 2013

Guardando la informacion queue_log a MYSQL



Creamos la  TABLA

USE asterisk;
CREATE TABLE queue_log (
  id int(10) UNSIGNED NOT NULL AUTO_INCREMENT,
  time char(26) default NULL,
  callid varchar(32) NOT NULL default '',
  queuename varchar(32) NOT NULL default '',
  agent varchar(32) NOT NULL default '',
  event varchar(32) NOT NULL default '',
  data1 varchar(100) NOT NULL default '',
  data2 varchar(100) NOT NULL default '',
  data3 varchar(100) NOT NULL default '',
  data4 varchar(100) NOT NULL default '',
  data5 varchar(100) NOT NULL default '',
  PRIMARY KEY (`id`)
);

martes, 12 de noviembre de 2013

Configuracion del SIP TRUNK de TRICOM


[tricom]
username=ambiorixg12
secret=1Erdo61
remotesecret=1Erdo61
fromuser=8097143489 ; aquí debemos colocar el numero con el cual tricom validara nuestras llamadas.
type=friend
insecure=port,invite
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes
host=190.6.132.37
directmedia=no

martes, 5 de noviembre de 2013

Como poner una emisora de radio como musica en espera en tu Central Asterisk .



Si deseas poner una emisora como música en espera. o simplemente transmitir una emisora de radio desde tu tu central Asterisk.  Aqui les dejo la configuracion que  uso en mi central telefónica. pueden probar cualquiera de las 2 emisoras o cambiar el link pero deben asegurarse que el link este en un formato que  mpg123 pueda reconocer y reproducir. Un punto imporante debemos tener instalado el mpg123.
 apt-get install mpg123

[default]
mode=custom
application=/usr/bin/mpg123 -q -r 8000 -f 8192 -s --mono http://radio2.domint.net:8022
;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -s --mono http://radio3.domint.net:8062

domingo, 20 de octubre de 2013

Codec G722 HD Audio

G.722 HD Audio. What’s the big deal?

G.722, HD Voice and HD Audio have become the latest buzzwords in the VoIP (Voice Over Internet Protocol) market in the last year. They are all words to describe the same thing - wideband audio that delivers voice calls using VoIP with audio quality that is greatly superior that of a regular landline or mobile phone call.  

sábado, 19 de octubre de 2013

Asterisk TLS Realizando llamadas seguras.

 

 Overview

So you'd like to make some secure calls.
Here's how to do it, using Blink, a SIP soft client for Mac OS X, Windows, and Linux.
These instructions assume that you're running as the root user (sudo su -).

Part 1 (TLS)

Transport Layer Security (TLS) provides encryption for call signaling. It's a practical way to prevent people who aren't Asterisk from knowing who you're calling. Setting up TLS between Asterisk and a SIP client involves creating key files, modifying Asterisk's SIP configuration to enable TLS, creating a SIP peer that's capable of TLS, and modifying the SIP client to connect to Asterisk over TLS.  

miércoles, 16 de octubre de 2013

Como cambiar el modo de una tarjeta digital de T1 o E1 sin move el jumper de configuracion.



The recommended way to set line mode on your Digium 1-, 2-, and 4-port (span) digital telephony cards is to set the jumper(s) on the card for either T1 or E1 mode for each span on the card. With the jumper off, the span is ready for T1 mode; with the jumper on, the span is ready for E1 mode. For more details about the jumpers, see the user manual for the single, dual, or quad span digital cards.
However, sometimes a card will have been installed in a server without first setting the jumper(s) correctly, and it may be inconvenient to remove the card from the server to access its jumper(s).  In this case, the "default_linemode" option can be passed when the card's device driver is loaded.  For single-span cards, the device driver is wcte12xp; for dual-, quad-, and 8-span cards, the device driver is wct4xxp.
Note: The eight-span card (TE820) does not have jumpers. The "default_linemode" module parameter, for the wct4xxp device driver, is the only method to set the TE820 line mode.
Depending on how your Asterisk server loads the DAHDI device drivers, the "default_linemode" option must be set/passed in one of two places:
  • If your Asterisk server uses the DAHDI init script to load the DAHDI device drivers on boot (or if you execute the "service dahdi start" command, which runs the init script), you must set the "default_linemode" option in the /etc/modprobe.d/dahdi.conf file.
    /etc/modprobe.d/dahdi.conf

martes, 15 de octubre de 2013

DAHDI











Short for "Digium Asterisk Hardware Device Interface"

The Change

DAHDI is the new name for 'Zaptel' as of May 19th 2008.
The post at http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/ details the reason for the change. Asterisk 1.4 releases later than 1.4.21, and all releases of Asterisk 1.6, will automatically use DAHDI in preference to Zaptel, even if Zaptel is still installed on the system.


Details should be available at http://www.asterisk.org/zaptel-to-dahdi  

viernes, 11 de octubre de 2013

Asterisk Dialplan Injection.



Hace una semana Olle Johansson anunció un fallo de seguridad bastante interesante, pero no me atreví a escribir sobre él hasta que no lo hubiésemos probado y al fín lo hicimos, y los resultados son escalofriantes:
Imaginemos que utilizamos un terminal IP (o softphone) con una cuenta limitada a extensiones SIP, en principio sólo podríamos llamar a extensiones SIP, pero el bug explica cómo aprovechar una mala programacion del dialplan y poder llamar a donde queramos:
El fallo de seguridad ocurre principalmente si tenemos una línea como esta:

Use asterisk to dial outbound number with extension


Sometimes, you want to reach an outbound number and when it answers, press some digits to naviguate through IVR.
The most common case where this can happens is when you want the callers from your PBX to reach a number and go direclty to a given extension. This prevents the caller to enters himself the extension number of the foreign side.
To make it done, you can use asterisk Dial option D.

domingo, 29 de septiembre de 2013

TE110P Jumper Configurations

TE110P Jumper Configurations

The Wildcard TE110P can be configured to support T1 (24 Channel), E1 (32 Channel, and J1 (24 Channel) by changing the card's jumper settings, listed below.
Jumper Settings
  • T1/J1 Jumper Settings — Jumper / OFF /
  • E1 Jumper Settings — Jumper / ON /
Back to Documentation
digium | asterisk

miércoles, 25 de septiembre de 2013

Como interconectar 2 centrales Asterisk usando troncales T1

 

How to connect two Asterisk PBXs using a T1 Trunk

This webpage will discuss how to connect two Asterisk PBXs together using a T1 trunk. The T1 trunk will be configured as an ISDN PRI rate trunk.
Brief Talk about T1 trunking
A T1 trunk consists of 24 channels of 64 kbps of data operating at 1.544 Mbps. It is a serial connection over UTP and provides full duplex communications. The connectors are RJ45 connectors the same as Ethernet. The pinout is different and a crossover cable pinout can be found here. The assumption at this point is that your T1 card has been tested and appears to be functioning properly.
The data channels can be configured to pretty much anything you want it to be. You can combine the channels to create a channel with 384 kbps bandwidth or you can use only part of the 24 channels. If you use only part of the T1 available bandwidth, it is called fractional T1.
The control of the T1 channels can be in-band or out of band. If it is in-band, bits are robbed from the 64 kbps channel for signaling. The result is 56 kbps for data per channel but you have 24 channels available for data. This robbed bit signaling is not used too much anymore due to the lower bandwidth per channel.
Out of band signaling uses one complete channel for control. This means that 23 channels of 64 kbps are available for data and the 24th is used for signaling. Any channel can be used for control, it doesn't have to be the 24th.

martes, 24 de septiembre de 2013

Astersik debug en un peer o IP especifica

1. The first method is invoked directly from the asterisk command line interface and allows to watch the output of the calls.

asterisk -rvvvvvvvvvvv
 sip set debug peer outbound-peer
 
This method will generate the sip debug for the peer that is specified, “outbound-peer”, to get a list of the peers run the asterisk cli command below:

sip show peers

sip set debug peer 100

 sip set debug peer callcentric

sip set debug ip 172.16.0.2

Asterisk sip show peer

root@ubuntu-PBX:~# asterisk -rx "sip show peers" | grep -i ok






root@ubuntu-PBX:~# asterisk -rx "sip show peers" | grep -i  un









jueves, 15 de agosto de 2013

DAHDI Timing Source


current versions of dahdi implement the fallback timing in the driver core,

martes, 13 de agosto de 2013

Sistema telefonico de consulta meterologica basado en Asterisk & Yahoo Weather


Aquí usaremos Asterisk & PHP  para crear un sistema de consulta del clima atreves de nuestra central telefónica. El Sistema verificara el Caller ID de la llamada y en base al código de área del numero que esta llamando nos dirá las condiciones climáticas para esa ciudad.

El sistema cuenta de  2 archivos en PHP:
YahoWeather.php
YahoWeather.class.php

También usaremos el TTS de Asterisk para que nos reproduzca el nombre de la ciudad y así evitar grabar los sonidos para los nombres de cada ciudad del mundo.

Continuación les dejo el código fuente.

sábado, 10 de agosto de 2013

Integrando el Servicio de Fax en Asterisk.


Benefits

  • Store faxes electronically
  • Reduce printing costs
  • Share faxes via email

Requirements

  • Server running Asterisk (32 bit compatibility needed)
  • Fax for Asterisk Software Add-on

lunes, 5 de agosto de 2013

Como enviar y recibir FAX usando ATA Grandstream.

This small Howto will show you a way how you can connect a Grandstream HT502 directly to a berofix device to use reliable Fax Transmission via T.38.
berofix: Creating a SIP-Account On the berofix Web-Gui you have to enter a new SIP account which points to the Grandstream device, as you can see in the next picture.
Bf sip account.jpg

viernes, 26 de julio de 2013

Asterisk time condition


[time]
 exten =>s,1,GotoIfTime(09:00-17:59,mon-fri,*,*?incoming-open,s,1)

;abrimos de lun-vi de  9 am a 5:59 pm

exten =>s,2,GotoIfTime(09:00-11:59,sat,*,*?incoming-open,s,1)

;también los sábados de  9 am -11:59 am

exten => s,3,Goto(incoming-closed,s,1)

viernes, 31 de mayo de 2013

Custom ringtone on calls from queue




Postby malcolmd » Fri May 31, 2013 6:58 am 


Code:
exten => 200,1,NoOp()
same => n,Answer()
same => n,SIPAddHeader("Alert-Info: <ringding>")
same => n,Queue(sales-queue)


Asterisk 11.4-ish

I dial 200, I hear my ringding ringtone, which is not the default ringtone for my phones, on each of my called phones. It works for both permanent Queue members (defined statically in queues.conf, e.g. SIP/101), as well as login/logout members who've logged themselves in via the Queues app on the phone.
 

miércoles, 8 de mayo de 2013

How to remove Asterisk


it is sometimes necessary to completely remove Asterisk for one machine, for example because you need to install a newer version.

Stop Asterisk and unload its modules
The first thing you have to do is to stop Asterisk and unload the modules it may be using, e.g Zaptel's.
The following lines will brutally terminate Asterisk and kill all ongoing conversation. You have to kill safe_asterisk first, otherwise it will respawn Asterisk.
killall -9 safe_asterisk
killall -9 asterisk

miércoles, 24 de abril de 2013

Contar y Limitar la cantidad de llamadas en Asterisk

Supongamos que tenemos una T1 o Troncal SIP con una cantidad de  24 canales y a l vez tenemos 6 DID. Podemos  asignar  4 canales a cada DID  para así evitar que un solo  DID consuma todos nuestros canales  y exista un balanceo.

lunes, 22 de abril de 2013

Restringir las llamadas basada en pines de seguridad

 [mycontext]
;disa
exten=>*4,1,Authenticate(/etc/asterisk/premiun_pass.conf,a)
same=n,Disa(no-password,mobile)

exten=>*5,1,Authenticate(/etc/asterisk/local_pass.conf,a)
same=n,Disa(no-password,localphone)

viernes, 19 de abril de 2013

Asterisk stops responding to SIP devices if it loses Internet Access (DNS)


If Asterisk loses internet connectivity or DNS, it stops responding to all SIP devices and trunks, and all extensions lose connectivity. This bug has apparently been around since Asterisk 1.4, persisted through 1.6, and remains in 1.8

viernes, 12 de abril de 2013

Que informacion necesito de mi provedor telefonico para configurar mi T1.


you'll want to know the framing and the coding.?

<framing>
The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1
Note: "d4" could be referred to as "sf" or "superframe"
<coding>
The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1
E1's may have the additional keyword "crc4" to enable CRC4 checking

You'll want to know the signaling type - is it a PRI, is it E&M?

You'll want to know how many voice channels you've got. If it's PRI where your d-channel is.

If it's PRI, what type of PRI signaling is it.

If it's E&M, you'll want to know whether it's immediate, wink, or Feature Group D.

You'll want to know how many digits you get on dialed numbers.

martes, 9 de abril de 2013

sábado, 6 de abril de 2013

Asterisk Channels Live




Esta aplicacion hecha para Windows, te permite monitorear el estatus de tus agentes y telefonos  en tu central Asterisk. EL projecto tiene dos sitios webs ademas  es muy facil de configurar en plataformas como Elastix y  Freepbx Distro.

http://www.astchannelslive.com/index.html

http://sourceforge.net/projects/astchannelslive/

Asterisk Local Channels


In Asterisk, Local channels are a method used to treat an extension in the dialplan as if it were an external device. In essense, Asterisk will send the call back into the dialplan as the destination of the call, versus sending the call to a device.
Two of the most common areas where Local channels are used include members configured for queues, and in use with callfiles. There are also other uses where you want to ring two destinations, but with different information, such as different callerID for each outgoing request.

martes, 12 de marzo de 2013

Como escribir una base de datos desde el Dial Plan de Asterisk


Hay muchas formas para realizar la tarea describa  en el titulo  de  mas arriba y este es una muy sencilla pero efectiva.


[mysql-context]
exten=>*59,1,Answer()
exten=>*59,2,Set(idcaller=${CALLERID(num)})
exten=>*59,3,system(/usr/bin/mysql --user=root --password='19582' asterisk -e 'UPDATE users SET password="${idcaller}"  where username="kl"';
)
same=>n,hangup()


exten=>*60,1,Read(pass,agent-pass)
exten=>*60,2,system(/usr/bin/mysql --user=root --password='19582' asterisk -e 'UPDATE users SET password="${pass}"  where username="kl"';
same=>n,Hangup()

miércoles, 6 de marzo de 2013

Asterisk Standard Channel Variables

There are a number of variables that are defined or read by Asterisk. Here is a listing of them. More information is available in each application's help text. All these variables are in UPPER CASE only.
Variables marked with a * are builtin functions and can't be set, only read in the dialplan. Writes to such variables are silently ignored.

lunes, 25 de febrero de 2013

Analizando el Trafico SIP de nuestra con red TCPDUMP y WIRESHARK


Una solución es utilizar TCPdump para la captura de la señalización SIP guardando los paquetes en un archivo y luego importar el archivo en la versión desktop de Wireshark (para Windows, MacOS y Linux).
Primero se capturan los paquetes SIP con TCPdump:
tcpdump -i eth0 -n -s 0 port 5060 -vvv -w /tmp/captura

domingo, 24 de febrero de 2013

Como enviar Alertas via SMS usando Asterisk.






Aquí les dejo varios ejemplos prácticos de enviar una notificación via SMS, ante diferentes eventos :

Los mensajes no se enviaran directamente desde Asterisk sino atreves de un  SMS gateway en la Internet
Tenemos que registrarnos en dicho Servicio puedes usar : http://www.bulksms.com/   tambien https://www.clickatell.com/

domingo, 17 de febrero de 2013

domingo, 20 de enero de 2013

miércoles, 16 de enero de 2013

Colecntando informacion de Debug.

Collecting Debug Information for the Asterisk Issue Tracker

This document will provide instructions on how to collect debugging logs from an Asterisk machine, for the purpose of helping bug marshals troubleshoot an issue on https://issues.asterisk.org

PREREQUISITES

 Asterisk 1.4.30 or greater

sábado, 12 de enero de 2013

Actualizando Asterisk.

 

Updating Asterisk

If this is your first installation, you can skip ahead to the section the section called “Base Configuration”. If you’re in the process of updating your system, however, there are a couple of things you should be aware of.
When we say updating your system, that is quite different from upgrading your system. Updating your system is the process of installing new minor versions of the same branch. For example, if your system is running Asterisk 1.8.2 and you need to upgrade to the latest bug fix version for the 1.8 branch, which was version 1.8.3, you’d be updating your system to 1.8.3. In contrast, we use the term upgrade to refer to changes between Asterisk branches (major version number increases). So, for example, an upgrade would be going from Asterisk 1.4.34 to Asterisk 1.8.0.
When performing an update, you follow the same instructions outlined in the section the section called “How to Install It”.  

viernes, 11 de enero de 2013

Digium TE122 chanelization



/etc/dahdi/system.conf
# Global data
loadzone = us
defaultzone = us
span=1,1,0,esf,b8zs
e&m=1-12
fxsks=13-24
echocanceller=mg2,1-24
span=2,0,0,esf,b8zs
fxoks=25-48
echocanceller=mg2,25-48
span=3,0,0,esf,b8zs
fxoks=49-72
echocanceller=mg2,49-72
span=4,0,0,esf,b8zs
fxoks=73-96
echocanceller=mg2,73-96

Versiones de Asterisk


Asterisk Versions

There are multiple supported feature frozen releases of Asterisk. Once a release series is made available, it is supported for some period of time. During this initial support period, releases include changes to fix bugs that have been reported. At some point, the release series will be deprecated and only maintained with fixes for security issues. Finally, the release will reach its End of Life, where it will no longer receive changes of any kind.
The type of release defines how long it will be supported. A Long Term Support (LTS) release will be fully supported for 4 years, with one additional year of maintenance for security fixes. Standard releases are supported for a shorter period of time, which will be at least one year of full support and an additional year of maintenance for security fixes.
The following table shows the release time lines for all releases of Asterisk, including those that have reached End of Life.
Release Series Release Type Release Date Security Fix Only EOL
1.2.X 2005-11-21 2007-08-07 2010-11-21
1.4.X LTS 2006-12-23 2011-04-21 2012-04-21
1.6.0.X Standard 2008-10-01 2010-05-01 2010-10-01
1.6.1.X Standard 2009-04-27 2010-05-01 2011-04-27
1.6.2.X Standard 2009-12-18 2011-04-21 2012-04-21
1.8.X LTS 2010-10-21 2014-10-21 2015-10-21
10.X Standard 2011-12-15 2012-12-15 2013-12-15
11.x LTS 2012-10-25 2016-10-25 2017-10-25
12.x Standard 2013-10 (tentative) 2014-10 (tentative) 2015-10 (tentative)
13.x LTS 2014-10 (tentative) 2018-10 (tentative) 2019-10 (tentative)
New releases of Asterisk will be made roughly once a year, alternating between standard and LTS releases. Within a given release series that is fully supported, bug fix updates are provided roughly every 4 weeks. For a release series that is receiving only maintenance for security fixes, updates are made on an as needed basis.
If you're not sure which one to use, choose either the latest release for the most up to date features, or the latest LTS release for a platform that may have less features, but will usually be around longer.


https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

martes, 1 de enero de 2013

Usando Google text to Speeach en Asterisk


Esto es algo rápido que aun no lo he organizado pero trabaja de maravilla.

Lo primero es que tenemos que tener instalada las siguientes dependencias.

perl : The Perl Programming Language
perl-libwww : The World-Wide Web library for Perl
sox : Sound eXchange, sound processing program
mpg123 : MPEG Audio Player and decoder
format_sln : Raw slinear module for asterisk
Internet access in order to contact google and get the voice data.