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.

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

DAHDI Timing Source

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

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.

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

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.