SPA400 y diferenciar cada FXO para controlar la llamada. - asterisk-es | Grupos de Google

SPA400 y diferenciar cada FXO para controlar la llamada. - asterisk-es | Grupos de Google: "Ya tengo mi SPA400 registrado en asterisk...
pero como un solo trunk...

hay forma de controlar cada puerto indepedientemente.,... osea saber q
entro una llamada
por determinado puerto.... para acorde al puerto de origen poder
controlar su destino.

actualmente...

register => SPA...@192.168.60.133/SPA400

[SPA400]
type=friend
username=SPA400
host=192.168.60.133
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=no
qualify=yes
context=entrantes

-----------------------------------------------

exten => _X000,1,Dial(SIP/${EXTEN})
exten => _X.,1,Dial(SIP/${EXTEN}@SPA400,50)

[entrantes]
; IVR por defecto
exten => SPA400,1,Answer
exten => SPA400,2,Background(menu)
exten => SPA400,3,WaitExten(5)
exten => SPA400,4,Dial(SIP/1000&SIP/2000&SIP/3000&SIP/4000,50)
exten => SPA400,5,Hangup

exten => 1,1,Dial(SIP/1000,50,m)
exten => 2,1,Dial(SIP/2000,50,m)
exten => 3,1,Dial(SIP/3000,50,m)
exten => 4,1,Dial(SIP/4000,50,m)

La idea seria que lo q entre por el puerto 1 de spa400 se valla a la
extension 1000, 2 a 2000, etc.

A proposito el background no me esta sirviendo... no me envia ningun
sonido.. alguna idea ?

Responder"

Extensiones detras de router - asterisk-es | Grupos de Google

Extensiones detras de router - asterisk-es | Grupos de Google: "HORMAX
Ver perfil
Más opciones 26 nov, 12:05
De: HORMAX
Fecha: Wed, 26 Nov 2008 08:05:19 -0800 (PST)
Local: Mié 26 nov 2008 12:05
Asunto: Extensiones detras de router
Responder | Responder al autor | Reenviar | Imprimir | Mensaje individual | Mostrar mensaje original | Informar de este mensaje | Buscar mensajes de este autor
Situación:
1.- Oficina central con Asterisk, lineas PSTN y trunk SIP, mas varias
extensiones que denominaremos A. Red conectada a Internet a través de
un router con el puerto 5060 direccionado al Asterisk.

2.- Extensión externa en oficina B, conectada a un router sin puertos
abiertos.

3.- Extension externa en oficina C, conectada a otro router sin
puertos abiertos.

Extensiones B y C se registran correctamente en Asterisk.

Llamadas A->B y A->C funcionan correctamente.

Llamadas desde A, B y C a la PSTN y trunk SIP funcionan correctamente.

¡¡¡ PROBLEMA !!! Llamadas B->A, C->A y B->C se cortan a los
20 segundos de establecerse.

Parece un problema de canreinvite pero:

. canreinvite= no, en todas las
exten"

Asterisk-Java

Asterisk-Java: "Users: Asterisk-Java and Spring
Mapping AGI URLs to Spring Beans

Sajid Moinuddin has blogged about using Asterisk-Java with Spring. His article shows how to implement a BeanNameAwareAGIMappingStrategy that automatically maps AGI URLs to Spring beans without any additional configuration.

References:

* Sajid Moinuddin: Asterisk-Java and Spring

Tags : asterisk-java spring
Add a comment RSS feed for responses to this blog entry
Posted by Stefan Reuter on November 13, 2008 2:50:00 AM CET #
AACC: An Asterisk-Java based solution for Call Centers
By: Fernando Borcel – AACC Project Manager

As far as I know, no one has yet succeeded to deliver a high quality open source package for the thousands of small and medium sized call centers around the globe that want to go Asterisk. But having to buy proprietary software for managing the call center, or having to build their own software, seemed up to now the only two available options.

AACC was conceived to fulfill the need of those call centers, and make an impact in call centers similar to the one Asterisk has made in telephony. Now, that’s setting a pretty high mark for ourselves, so, how will we do it?

First of all, we have tried very hard to understand what those small and medium sized call centers need and want. So we are designing a system which is feature rich, yet easy"

Avisar cuando una extensión no existe - asterisk-es | Grupos de Google

Avisar cuando una extensión no existe - asterisk-es | Grupos de Google: "Avisar cuando una extensión no existe
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marcar
8 mensajes - Contraer todos
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melfree
Ver perfil
Más opciones 23 nov, 01:01
De: melfree
Fecha: Sat, 22 Nov 2008 21:01:20 -0800 (PST)
Local: Dom 23 nov 2008 01:01
Asunto: Avisar cuando una extensión no existe
Respo"

Ubuntu Linux stop / disable GNOME GUI ~ X.org

Ubuntu Linux stop / disable GNOME GUI ~ X.org: "Ubuntu Linux stop / disable GNOME GUI ~ X.org
Posted by Vivek Gite [Last updated: September 7, 2007]

Q. I don’t need GUI as I’m developing CLI based applications. By default Ubuntu Loads Gnome GUI. How do I disable X.org / Gnome under Linux so that I get text only login?

A. You can use GUI tools or command line tools to disable GDM (Gnome Display Manager) service (/etc/init.d/gdm).
Disable GDM using GUI tools

The Services Administration Tool allows you to specify which services will be started during the system boot process. You can type the command:
services-admin &

Or just click on System > Administration > Services

Now you will be prompted for the administrator password, this is necessary because the changes done with this tool will affect the whole system. After entering the administrator password, the following window is displayed:

Ubuntu Linux stop / disable GNOME GUI (X.org)

Make sure you remove GDM (Gnome login manager) by disabling the the checkbox and close the window.
Enable GDM using Command Line (CLI) tools

Ubuntu comes with rcconf and update-rc.d command. rcconf allows you to control which services are started when the system boots up or reboots. It displays a menu of all the services which could be started at boot. The ones that are co"

» Instalación de Asterisk 1.4. y la Asterisk-GUI :: SinoLogic :: Noticias y comentarios sobre VoIP, Asterisk, PBX, GUI, SIP, Unified Communications, etc...

» Instalación de Asterisk 1.4. y la Asterisk-GUI :: SinoLogic :: Noticias y comentarios sobre VoIP, Asterisk, PBX, GUI, SIP, Unified Communications, etc...: "Instalación de Asterisk 1.4. y la Asterisk-GUI

GUIDespués de la noticia que publiqué hace un par de días sobre la nueva interfaz web de Asterisk, he recibido unos 20 emails preguntándome cómo se instala. Incluso Alberto Sagredo (VoipNovatos.es) ha puesto un post sobre cómo instalar Asterisk 1.4 y la interfaz web.

Lo primero que quiero comentar es que la Asterisk-GUI, sólo funciona en Firefox y Ópera (No Internet Explorer), cuando los de IE acepten seguir las normas W3, entonces será compatible.

Para instalar Asterisk 1.4 es tan sencillo como (distribuciones basadas en Debian):

cd /usr/src

apt-get install -qy bison flex libncurses5-dev zlib1g-dev libssl-dev libnewt-dev libiksemel1-dev

apt-get install -qy make linux-headers-`uname -r` gcc g++ libstdc++6

(apt-get install cualquier otra librería o paquete que creais que falta… no todas las distribuciones son iguales…)

wget -c http://ftp.digium.com/pub/zaptel/releases/zaptel-1.4.0-beta2.tar.gz
wget -c http://ftp.digium.com/pub/libpri/releases/libpri-1.4.0-beta1.tar.gz
wget -c http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0-beta3.tar.gz
wget -c http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.4.0-beta2.tar.gz

tar xvfz zaptel-1.4.0-beta2.tar.gz
tar xvfz libpri-1.4.0-beta1.tar.gz
tar xvfz asterisk-1.4.0-beta3.tar.gz
tar xvfz asterisk-addons-1.4.0-beta2.tar.gz

cd zaptel-1.4.0-beta2
make && make install
cd ..

cd libpri-1.4.0-beta1
make && make install
cd ..

cd asterisk-1.4.0-beta3
./configure && make && make install && make samples
cd ..

cd asterisk-addons-1.4.0-beta2
make && make install
cd ..

Con esto ya tendríamos el Zaptel, Libpri, Asterisk y Asterisk-Addons instalado, ahora nos falta la parte fácil:

apt-get install -qy subversion

svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui

cd asterisk-gui

make && make install && make samples

cat < < EOF >/etc/asterisk/http.conf
[general]
enabled=yes
enablestatic=yes
EOF

cat < < EOF >/etc/asterisk/manager.conf
[general]
displaysystemname = yes
enabled = yes
webenabled = yes
port = 5038
;httptimeout = 60
bindaddr = 0.0.0.0

[admin]
secret = mypass
read = system,call,log,verbose,command,agent,config
write = system,call,log,verbose,command,agent,config
EOF

make checkconfig

cd ..

asterisk -gvvvvvvvvc .

y mientras tanto, escribis en vuestro navegador Firefox o bien Ópera (… recuerdos a Saghul xD):
http://DIRECCION_IP:8008/asterisk/static/config/cfgbasic.html

Os autentificais con usuario: admin y contraseña: mypass

y listo!

Ranking de GUIs - asterisk-es | Grupos de Google

Ranking de GUIs - asterisk-es | Grupos de Google: "No voy a valorarlas, pero puedes probar las siguientes:
VoiceOne,
Druid,
Asterisk-GUI,
FreePBX,
RaGUI,
Elastic
..."

VoipJots.com: Click-to-Call with your Asterisk@Home Box

VoipJots.com: Click-to-Call with your Asterisk@Home Box: "Click-to-Call with your Asterisk@Home Box


It seems everyone is talking about Click-to-Call functionality these days. Google has announced they are going to do it, EBay will be doing it before too long, and there are other websites out there that I've recently noticed are offering this service.

Click-to-Call seems like it would be a complicated task to setup; enabling a website visitor to click a button to allow them to submit their phone number, and receive a call from someone a few seconds later, with no long distance charges for the visitor.
If you have an AAH or Asterisk box, it really is not that difficult to add such a feature to your website. A very simple PHP page hosted either on your AAH box, or another PHP-capable web server (that has TCP connectivity to your AAH/Asterisk box) is all you need. Last night I developed such a script, and though it is rather simple, it's effective. I plan on doing more with it later, but thought I'd share it will all of you so that you can begin playing with it, or customizing it yourself. This script was written for AAH, so it probably will not work with a regular Asterisk box without some modifications, or even a heavily customized AAH box. However, it's very straight-forward, and should not be hard to debug if you have problems with it."

To the expert PHP'ers out there, this was my first attempt at doing anything with PHP (I'm an ASP guy), so have pity on me!:)

Ok, here it goes.

I'm going to be explaining how to set this script up on an AAH box, but as I said before, it should work on any PHP server that can communicate with your AAH/Asterisk box. Also, I have only tested this on AAH 2.4, so I don't make any guarantees that it works on anything else. In fact, I don't make any guarantees that it works on any AAH 2.4 box besides my own, though it should. :) While I'm at it, I may as well also mention that if you decide to use this script, you do so at your own risk.

Step 1:

Login to your AAH box and enter these commands:


cd /tmp
wget http://downloads.voipjots.com/scripts/click-to-call.zip
unzip click-to-call.zip
mv click-to-call.php /var/www/html/click-to-call.php
rm click-to-call.zip



Step 2:
Now we have to slightly edit the script to get it to work for you.


cd /var/www/html
nano -w click-to-call.php


For a typical AAH box, you should only need to edit one line in the script. Change the line:
:


$strChannel = "IAX2/250";


to reflect one of your extensions. In my case, I am using my Idefisk IAX2 softphone that has an extension number of 250. For instance, if you have a SIP soft/hard phone that has an extension number of 1234, you would change this line to:


$strChannel = "SIP/1234";



Ok, at this point, we should have everything properly configure on our AAH box. Now we can try it!

Step 3:

In a web browser, go to http://my-aah-ip-address/click-to-call.php

You should see this:




While near the soft/hard phone belonging to the extension you specified above, enter a valid 11 digit phone number, and click on the Call Us! button.

The soft/hard phone should begin ringing, showing "Web Call " as the caller id. When you answer the call, you will briefly hear... nothing (something I'm going to be working on in the near future), but after a few seconds, you will hear ringing as it calls the number that was entered on the web page.

The click-to-call web page should now be this:




This script is very simple, and probably not best hosted on your aah box, but on another web server that hosts the site you would want to add click-to-call functionality to. However, I hope that this provides a good demonstration as to exactly how simple it is to implement click-to-call with AAH, and leave it to you and your creativity on exactly how you wish to offer this feature. The sky's the limit, really.

I hope this has been helpful to someone. If you have any ideas on how to improve the script, please let me know by leaving your comments. If you make significant improvements to the script, I would appreciate it if you shared them with all of us.

Enjoy!
Olin

Implementing VLAN trunking

Implementing VLAN trunking: "Windows configuration with Intel Pro Series adapters
Conceptually, trunking a Windows workstation or server to a switch is the same a trunking a router to a switch. The only difference is the procedure, and a much easier one I might add. The ubiquitous Intel Pro Series adapters provide a simple to use graphical tool called PROSet that any one can learn within a minute or even someone who is just winging it. Note that the same Intel adapters with the ANS drivers can provide similar capabilities on Linux. You can get more information on Linux here from Intel.

To get started, simply invoke the Intel PROSet or PROSet II utility (assuming PROSet is installed). This can be done by simply double clicking the PROSet icon in the system tray on the lower right hand corner of the desktop. The following utility should come up.


Next we must add a VLAN interface. Simply right click on Intel adapter with the PCI Card icon and click �Add VLAN�. Note in the following screen capture, the virtual interface for VLAN 100 is already there and we are adding an additional one.


The �Add New VLAN� window comes up. Enter the VLAN ID you want this interface to trunk in to in the ID field, then give it a name that describes the VLAN function. In this case, we will be adding VLAN 69 labeled the W"