Honor to Bob Fryer from Australia, he wins the first prize!

Second place to the French forum moderator Franck Danard, followed by other ten great community members (like Ben Sharif, a great photographer and the author of Elastix Without Tears).

I must say thanks to these people. They helped me a lot during the last year to understand VoIP and Elastix PBX.

HERE the article.

Date of the event: 15-01-2010 (Stay tuned on www.elastix.org).

This is a nice idea, and it is the final confirmation of the good shape of the Community behind the project.

These are the prizes:

1. First place: US $1000
2. Second Place: US $500
3. Ten “Honor Contributors of the Year”: each one receiving an IP phone.

HERE you can view the sponsors of the Event.

After I got notification on the Elastix Beta mailing list about FreePBXv3, I was stoaked to learn it’s been released, even in an unfinished form! I went looking a while back for a FreeSWITCH GUI and was sadly disappointed. I found a few like tcapi and Phonebooth, but couldn’t get tcapi working and Phonebooth had just had all its code donated to FreePBX. So for the time I’d given up.

Anyway, so I grabbed the latest Install CD for FreePBXv3 which is based off CentOS. Ran through smoothly, no issues, got it installed, but I didn’t really do anything much with it.

For one reason or another I jumped into IRC in #freeswitch on freenode, I’ll have to look over my logs and figure out why. Anyway it turns out that some of the FreePBXv3 devs hang out there, and they’re actually friendly! It was a nice change from some of the other projects I’d had involvement in. I think Elastix is probably the only other project that’s been so friendly.

So I got chatting with one of the guys there, and it came up that I was happy to do some testing, so he asks me to do a clean install from a vanilla distro (And not the LiveCD. Again I’m not sure why, I’ll go re-read the chat logs later). No worries, so I install vanilla debian and get to work. Anyway a few bug reports later, 3 SVN updates later, and I’ve got a nicely working system running currently in VirtualBox.

I’m pleased to say that FreeSWITCH as a product from what I can see appears quite mature, but still growing at a steady rate. They even recommend you install from SVN, they’re that confident in the code. Most projects would have you install from a “stable” peg in the code and update as & when they release a new “final” version. The first and most obvious thing I’ve noticed about FreeSWITCH is the sounds. Even just the female voice that guides you through recording a Voicemail sounds *so* much nicer than the Asterisk one, it’s absolutely crazy! There’s also the nice feature that Voicemail automatically stops recording when you’re silent for 2 seconds. Quite cool I thought!

There was no “additional” configuration of any endpoints that I needed to do, tested with ZoIPer and an SPA942 successfully. It simply “just worked” making calls between the devices, as was to be expected. I’m still yet to setup an external trunk to test over.

FreePBXv3 is still a *little* rough around the edges, but overall I must say that the infrastructure that they’ve built, or framework, seems like a great improvement over the current FreePBX 2.X! Not only that, but the developers are very approachable, keen for feedback, they seem to be excellent coders who love doing what they do, and they’re rapidly pumping out an awesome project! It still crashes at times, or at least doesn’t display what it should, however the changes seem to get saved at least even though the correct followup output isn’t shown. Some features aren’t 100% implemented, but overall there’s most of the basic things in there that the majority of people would use. The developers also seemed pretty keen for suggestions and improvements, even just the trivial little things.

I’m just looking forward to trying this out on an Alix system soon to see how it handles the load from FreeSWITCH

On 31 December 2009 Elastix.org have published some previews about Elastix 2.0 PBX Server… Nothing to say, there are great news in the next future for our users of the supreme PBX System based on CentOS.

Elastix 2.0 Dashboard

Here, you can see the article…

If you have some free time, my advice is to test Elastix giving a great help during the debug time (this is the Alpha 3 version of the 2.0 system).

To Download: 32 bit –  64 bit

To Help with Debug:  Bug Tracker

Enjoy people, enjoy…

Happy 2010 again to you all…

However, it comes at a price, 64kbps + overheads means around 111kbps when you factor in everything else over an ADSL PPPoA / PPPoE connection.

Now that’s a LOT when you think about it, considering on a standard ADSL2+ line you’re going to max out at around 5-7 SIP lines, especially if it’s a shared connection. This is where a compressed codec such as (My personal favorite) iLBC, or g729, can be incredibly cost effective, as you can load up around 15-20 on the same sort of bandwidth. When you’re a small business, that means with the right kind of QoS, you can share your ADSL Broadband connection and still have 5-10 concurrent phone calls, all happily living together.

So do away with expensive BRI interfaces and monthly charges, and go SIP!

We’re going to look at a few things very briefly:

1) MOS – What is it and why do I care?

2) Which codec is right for me?

3) g729 – Installation

4) iLBC – Installation

5) Trunk and Extension setup

6) Testing the codecs

1) MOS – What is it and why do I care?

MOS stands for “Mean Opinion Score“, and in a nutshell it’s a way of determining out of 5 how good a particular codec (Read: Phone call) will sound. The Wikipedia article is a good overview of MOS, each codecs ratings, and some phrases you can use to test the compression yourself and see if words become difficult to understand.

Now, we all want a nice good MOS, because at the end of the day if you’re using a PBX system, there’s nothing worse than it sounding like a VoIPBuster call to some exotic island. Personally I aim for higher quality, this is why iLBC stands out well for me. That said, the choice is yours, try a few out and see what sound you personally (And quite possibly the person who writes your paycheck) are happy with.

2) Which codec is right for me?

Well that is a very good question, and that’s something that you’re going to have to answer for yourself. Lets look at some of the benefits of each:

It’s not difficult to see there could be benefits for either. If you’re looking for low-cost deployment, or over a shaky ADSL connection you can’t seem to QoS well (Or perhaps Wireless?) then iLBC is probably the way to go. If you’re doing a larger installation, or an installation for a customer then it’s worth investigating g729, especially if you’re running underpowered hardware or have a large number of Exts or lots of expected concurrent calls.

Please keep in mind that you do not have to stick with just one codec, feel free to mix and match box on a per-customer level, or on a per-machine level also.

3) g729 – Installation

OK so I’m going to cheat here and direct you to Digium. Yes, there are “free” binaries out there, but at the price of $10 per-channel surely it’s not that expensive if you’re going to add it to your PBX’s you deploy.

Anyway if you’re downloading a “free” version, simply pop it in /usr/lib/asterisk/modules and chmod 755 the file. 64-bit versions go in /usr/lib64/asterisk/modules

Other than that, follow Digiums excellent instructions here, it should take you less than 5 minutes: http://kb.digium.com/entry/5/

4) iLBC – Installation

We’re going to be compiling this from source. Lucky for us Elastix makes this nice and easy! We won’t be installing a complete asterisk system, we don’t want to break the RPM’s because, let’s face it, it’s the easiest way to keep a system updated / secure.

First off, download Asterisk from here: http://www.asterisk.org/downloads

Get it to your elastix box any way you want (SCP / SSH / local terminal) and pop it in /usr/src

Now, run:

tar xvzf asterisk-1.4.*.tar.gz

change in to the newly created “asterisk” directory and run:

./contrib/scripts/get_ilbc_source.sh

Press “Enter” after it displays the warning presuming you’ve read the documentation (Which I’m also going to assume you’ve done) and shortly after you’ll get a lovely message telling you “The iLBC source code download is complete.”

Now, run:

./configure

That should run through without any issues, so now we’re going to tell Asterisk that we only want to build a very basic system (for the sake of time) and we want the iLBC codec. To enable the iLBC Codec you need to ensure in “4. Codec Translators” that “codec_ilbc” is checked.

Note: When you press “Esc” to quit “make menuconfig”, make sure you press S to save changes.

Now, just type “make” and watch the system build! As soon as you see codec_ilbc you can press Ctrl + C to stop it building. You’ll notice codec_ilbc being built just after all the app_ files, so it’s safest to stop the build process when you see the format_ files being built. Either that or let it run right the way through.

Almost done now, you just need to copy ./codecs/codec_ilbc.so to /usr/lib/asterisk/modules (or /usr/lib64/asterisk/modules on a 64-bit system) and chmod 755 /usr/lib/asterisk/modules/codec_ilbc.so

5) Trunk and Extension setup

Now depending on if you want your Ext or SIP Trunk to use the codec, it may be desirable to “force” the system to only use that codec.

To setup a Trunk: Open the SIP / IAX2 Trunk that you wish to limit in Elastix to the particular codec. In the PEER Details add an additional two lines:

disallow=all
allow=g729&ilbc

You may want to opt for something such as “allow=g729&g711″ or perhaps “allow=ilbc&alaw” for example. Review your needs and act appropriately. Save and submit the changes.

To setup an Ext: Open the Ext that you wish to limit the particular codec on. Scroll down to “disallow” and enter in “all”. Next in the “allow” field enter in something such as “g729″ or “ilbc”, or perhaps “ilbc&alaw” as above with the Trunk. Save and submit the changes when done.

Notes: Try and avoid transcoding where possible. It’s nice that asterisk does it so well, however it does add marginal delay to your calls, not to mention CPU overheads etc.

Also keep in mind that if you limit the codecs, say to just g729, and a particular device is unable to use g729, the call will simply fail.

6) Testing the codecs

The best part, testing! Hopefully everything has gone according to plan thus far. You’ve got one or two new codecs setup and you’re dying to test them out (In a development / testing environment naturally and not on your companies production PBX). Now you need to ssh in to your asterisk system and run:

asterisk -r

This brings up the asterisk console. Now, make a call on one of your new Ext’s that should be using iLBC / g729, or over the trunk that should be compressed.

While that call is active, rush back over to your ssh session and in the asterisk console type:

sip show channels

You should see something like this:

x.x.x.x   09xxxxxxx   2408517b625  00101/00102  0x400 (ilbc)     No       Rx: ACK

If you’re making a pure SIP call, you’ll see one line for the Ext -> Asterisk, and one line from Asterisk -> your SIP Provider.

Success!

Thanks for reading, and if you have any questions please feel free to leave comments or post in the Elastix forums.

Please also keep in mind that whilst g729 / iLBC are not the only codecs, almost every device that supports compression is likely to support one of them, or if not it will support GSM.

Hopefully as you’re reading this you’re happily making calls via iLBC or g729. Enjoy!

Cost savings, flexibility, functionality, and I’ll say it again: Cost savings!

When deploying Elastix to use SIP over ADSL (for example), many find their existing Broadband connection does not quite provide satisfactory call quality, usually due to sharing the connection with other traffic.

We will follow this post up later with another on Diagnosing connectivity / quality issues.

NOTE: This is not the only way to do QoS, but after having struggled with the likes of pfsense, the budget / useless junk that many routers build in, and a host of other software / hardware solutions, I found Tomato did it the easiest, the best, and the most reliably! It’s a breeze to setup, and you’ll be kicking yourself for not having set something like this up earlier.

Requirements:

• Administrative access to the ADSL Router
• The Router must either do DMZ or (most ideally) Half-bridging. The Linksys AM300 is recommended
• A Linksys WRT54GL or Asus WL-52oGU — The WRT54GL was easiest to flash, but the WL-520GU is un-brickable
• The Tomato Firmware, available from polarcloud.com/tomato
• A quiet afternoon in the office, where nobody will jump up and down too much if you take down the internet for a little bit.

NOTE: I would *strongly* recommend you get permission to take and test this out on your home connection prior to trying to deploy it in a production environment. I promise you will suffer much less heartbreak compared to tearing down a corporate network and finding you cannot easily repair it to its original state.

Step 1) Selecting the Hardware

I’ll be honest, the Linksys WRT54GL is probably the easiest, and at around NZD$100 it’s a bargain price! See xe.com/ucc to convert that price to your currency, or check out the price at Newegg.com

Make sure its the GL model, the L standing for Linux, as some of the non-L models aren’t compatible, namely the WRT54G v5.

You also need a router that will hand the IP address / port-forwarding over to it, so you’re going to want something such as the Linksys AM300, it’s cheap, reliable and known to work well. Just make sure it has Firmware 1.19.04 or half-bridge won’t work. Other products such as the NetComm NB6PLUS4Wn will suffice if you put the Tomato in a DMZ, but half-bridging is a nicer solution than a DMZ, due to the Tomato router thinking it’s pretty much *the* connection to the Internet. I’ll leave the getting-connected-to-the-internet part to your imagination.

Step 2) Flashing Tomato

If you have the WRT54GL then this is dead easy. If you’ve previously used it, then reset it back to factory defaults. Connect via a cable (safer than wireless, even though technically speaking WiFi does work for upgrades), go into the Admin settings, select the firmware: WRT54G_WRT54GL.bin

Hit Upgrade, sit back, wait 2-3 minutes and then try re-logging in to the Web Interface.

devices such as the Asus WL-520GU also work well in my experience, though the WRT54GL is definitely the easiest.

Now there is a ton of features that we could go into which you may find useful, but now’s not really the time to go into all those. With that in mind we’re going to dive straight into the deep end. Fire up the WebGUI, Login with the default username:password of admin:admin and click on QoS –> Basic Settings on the left-hand side.

Step 3) Determining your speeds

This may sound like a silly thing for me to mention, but you absolutely must get this right. First stop: Speedtest.net !

Run a speedtest and see what speeds you get. Also if you check the sync speeds in your router, they’ll give you a rough indication of what speeds you’ll be getting.

Now, you’re not going to want to set your Max InboundBandwidth & Max Outbound Bandwidth in Tomato to your actual line max. For example if your DSL connection is sync’ing at 4.8m/bit down and 950kbps upload and you specify those values in Tomato, not only will you never actually attain those speeds, but if you re-sync at a slightly lower speed, it’ll throw things even more. Speedtest for me shows I’m getting around 4300kbps down and 890kbps upload, so we’re going to round things right down to 3500kbps download and 800kbps upload. Seriously, I can pretty much promise that none of your staff will notice the difference between the internet speeds downloading at 4300kbps vs 3500kbps so don’t feel as though you’re going to be slowing down their internet. You won’t! If anything, they’ll “feel” like it’s going faster by the time we’re done.

So, enter in those values of yours in the Max Bandwidth for inbound & outbound. Up the top we’re going to set:

• Enable QOS
• Prioritize small packets with these control flags: ACK
• Prioritize ICMP
• Reset class when changing settings
• Default Class: C

Your Default Class can change, I’ve set it to C for un(C)lassified. Honestly it made sense when I first setup the router…

Now, we’re going to set the Outbound values, these are the most essential. Here’s a bit of an example of how mine looks:

• Highest / 80% / 90%
• High / 20% / 80%
• Medium / 5% / 70%
• Low / 3% / 60%
• Lowest / 2% / 20%
• Class A / 90% / 100%
• Class B / 1% / 5%
• Class C / 1% / 10%
• Class D / 1% / 5%
• Class E / 10% / 20%

Now I’ll explain my madness here. Class C is the default, so if it’s uncategorized it’s because it’s not important enough for me to spend the time writing rules for it, so you don’t want to give it much. Class B & D don’t get used by me, so again I leave them very low, as nothing should be using them. Class A is going to be my VoIP so I want it to always get priority.

Here’s a summary of my understanding of how the QoS works. It could be totally wrong, however it’s done me well thus far:

You’ve got a limited amount of outgoing bandwidth, lets say for arguments sake its 100kbps.
Now lets pretend you’re saturating that with a variety of traffic, including VoIP, sending email attachments through Gmail (http traffic) and FTP uploading (website changes?). So, VoIP is only going to use a tiny little bit, if you’re using iLBC then it’ll be 13.3kbps + overheads = 44kbps (See here). If you’ve classified VoIP to Class A, it’s going to try and reserve 90% at least. You only need 44kbps so that’s still leaving 66kbps for HTTP & FTP. Now, if HTTP is Highest, then it’s going to try and reserve 80% of the 100kbps for it, essentially leaving FTP with nothing. For this reason you’d set HTTP to High perhaps, and FTP to Medium. Both have a “max” of 80% & 70% respectively, so it’ll do it’s best to share the remainder after VoIP between the two of them, at all times giving HTTP a minimum of 20% and FTP at least 5%.

In a nutshell, set the minimum amount to be reserved on the left, and the max that it should be allowed on the right. Assume worst-case scenario that your line is going to be 100% full all the time, and try and give each a bit of “wiggle room”

Next we setup Inbound also:

• Highest / 90%
• High / 80%
• Medium / 70%
• Low / 60%
• Lowest / 20%
• Class A / 100%
• Class B / 5%
• Class C / 10%
• Class D / 5%
• Class E / 5%

You can turn on TCP Vegas if you want, I choose not to, you can read more about it and what it does on Wikipedia. Scroll down and hit Save.

Now on to the next part:

Step 4) Classification of Traffic types

On the left-hand side we’re now in QoS –> Classification

We need to classify the types of traffic that fall in to each of the classes we setup just earlier. We’re going to start by going through and clearing out all the standard rules and we’ll put in a few of our own:

• UDP / Src or Dst / 5060 / Class A / SIP
• UDP / Src or Dst / 10000-20000 / Class A / RTP
• TCP or UDP / Dst Port / 53 / Highest / DNS
• TCP / Dst Port / 80,443 / High / HTTP & HTTPS
• TCP / Src or Dst / 25,110,143 / Medium / Email (SMTP POP3 IMAP)
• TCP or UDP / Src or Dst / Class C Bulk Traffic
• GRE / Src or Dst / Class E / PPTP VPN (GRE)

Why? We prioritize VoIP first of all, at all times we want our voice to be perfect. Next we do DNS so that pages “feel” like they’re loading quickly (Especially with OpenDNS). We want HTTP & HTTPS in their own little queue, followed by Email.
Im also putting PPTP VPN into a class of it’s own, as I use that a bit myself for a variety of things.
Finally, we’re already technically classifying anything that isnt classified as Bulk Traffic / Class C, but this rule just re-affirms it.

Scroll down, hit Save, then make a call to a local freephone number out your local SIP connection to see if this is working.

Step 5) Confirmation and Corrections

Now comes the testing part. You should be able to click on QoS –> View Graphs, and while you’re on a call you’ll see Class A has a bit of traffic in it. If you click on the Class A title it will take you to a screen where you can now also see the IP that the traffic is going to. With a bit of luck, this should be your Elastix box (If you’ve got a remote extension) or your ITSP (If your Elastix box is local).

From here, it’s a matter of keeping an eye on the graphs over the next few days / weeks and seeing what really goes on through your Internet Connection. You may be quite surprised at what you see.

If you run into any troubles, feel free to post on the Elastix Forums and either myself or one of the many other helpful regulars will be able to assist.

This is an important news for all people like me that are studying (for hobby or work) the fantastic world of the Voice Over Internet Protocol, alias VoIP.

After one year in this business I can surely put Elastix Server (developed by Palosanto Solutions) on the top of the ladder about the PBX Servers based on Asterisk. It is reliable, stable, customizable, and followed by a huge community of developers and people that well know this business.

The new effort from this community is this Blog. It is updated very often by two veterans from the Elastix’s Forum… Bob (alias Bob Fryer) from Australia and Chilling_Silence from New Zeland.

These guys helped me a lot during the last year… And now they decided to help all people interested in the Project.

They will share all their experience on the blog, but there is more… Anyone who wishes to submit an article, can send it to the following E-Mail address rbonifaz[at]elastix.org (he is the Rafael of the Forum, from Palosanto Solutions).

The guys have already posted some interesting stuff… Check It Out!!!

AstNext specializing in telecom Solutions, has come to be one of the industry’s leading solutions provider companies. Since 2006, AstNext Telecom Solutions has been providing total solutions to our customers, creating innovative, bullet-proof communication solutions. With a product line ranging from EPABX, IVR, Predictive dialers, Voice Mail, Voice Logger, Video & audio conferencing solutions, web conferencing solutions and many more customized telecom solutions, AstNext guaranteed to have a product to suit your communication needs.

Technical Details:

“Quito, Ecuador. El pasado Domingo 26 de Abril de 2009 se realizaron elecciones generales para presidente y gobiernos seccionales en el Ecuador. Para este proceso electoral se decidió optimizar la comunicación en todos los lugares donde se realiza el conteo de votos para monitorear el proceso. Elastix fue elegido como plataforma de comunicaciones por sus beneficios técnicos y económicos para el proyecto.

Elastix pudo comunicar sin inconvenientes a mas de ochenta (80) localidades a nivel nacional para conteo de votos. Gracias a Elastix se pudo implementar un sistema de comunicaciones de gran envergadura a nivel nacional en tan solo dos (2) semanas!

Por medio de Elastix se pudo proveer funcionalidad avanzada de call center para ayudar a solucionar problemas técnicos referentes al proceso Electoral.”

Queda demostrado nuevamente lo confiable, estable y  robusto que es una Central Telefónica Asterisk, también la capacidad que tiene para usarse en instalaciones de gran tamaño como lo son: Universidades, ministerios, Hospitales, Hoteles, etc.

También tenemos que felicitar a los amigos de PaloSantos por desarrollar un sistema en nuestro idioma.

Sigan Adelante

Saludos

[ctorres@asterisk]# mkdir /respaldos

vamos a trabajar con el cliente ftp ncftp, lo bajamos e instalamos de la Siguiente manera

[ctorres@asterisk]# cd /usr/src
[ctorres@asterisk]# wget ftp://ftp.ncftp.com/ncftp/ncftp-3.2.2-src.tar.gz
[ctorres@asterisk]# gzip -d -c ncftp-3.2.2-src.tar.gz | tar xf -
[ctorres@asterisk]# cd ncftp-3.2.2
[ctorres@asterisk]# ./configure
[ctorres@asterisk]# make
[ctorres@asterisk]# make install

Luego creamos el batch de respaldo (esto tambien funciona bien para elastix)#!/bin/sh

[ctorres@asterisk]# nano backup.sh

y le pegamos el siguiente codigo
#!/bin/sh

echo &(date +%d/%m/%Y-%T) | mail -s “Respaldo Central ” sucorreo@mail.cl

cd ~/

mysqldump -uroot -peLaStIx.2oo7 –opt asterisk -C -c> /respaldos/asterisk.sql
mysqldump -uroot -peLaStIx.2oo7 –opt asteriskcdrdb -C -c> /respaldos/asteriskcdrdb.sql

cd /respaldos
tar -zcvf asterisk_$(date +%d%m%y).tgz *.sql
tar -zcvf etc_asterisk_$(date +%d%m%y).tgz /etc/asterisk/*.*
find -name ‘*.tgz’ -type f -mtime +2 -exec rm -f {} \;
rm *.sql

### Dump backup using FTP ###
#Start FTP backup using ncftp
/usr/src/ncftp-3.2.2/bin/ncftp -u”usuarioftp” -p”passftp” ftp.sitio.cl<<EOF
cd /asterisk/enea/
lcd /respaldos
mput *
quit
EOF

cd /respaldos

rm /respaldos/*.tgz

en la 1º linea en el echo enviamos un correo indicandonos cuando fue hecho el respaldo (cuando se ejecuta el script)

en mysqldump utlizamos el usuario root [-uroot] con la password eLaStIx.2oo7 [-peLaStIx.2oo7], [asterisk] seria la base de datos .

[ftp.sitio.cl] es el sitio ftp donde subiremos los archivos de respaldo.

[usuarioftp] es el usuario ftp

[passftp] es el password del usuario para el sitio.

luego le damos privilegios de ejecucion a nuestro script

[ctorres@asterisk]# chmod +x backup.sh

probamos nuestro script

[ctorres@asterisk]# ./backup.sh

si todo salio bien, vamos a programar nuestro script para que corra a las 01:00 todos los días.

[ctorres@asterisk]# crontab -e

insertamos la siguiente linea y grabamos (Esc y ZZ para salir en Centos)

0 1 * * * ~/backup.sh

con esto ya creamos nuestro batch de respaldo para las bases de datos (en este caso para nuestro asterisk).

Neowolfman

Miolo es un framework para la creacion de sistemas qeu son accesibles via WEB.

Edgar Landivar promotor y creador del proyecto Elastix a publicado en línea la versión beta del libro “Comunicaciones Unificadas con Elastix” bajo licencia GNU FDL. Este libro no asume por parte del lector conocimientos previos en telefonía, voz/ip, Linux o Elastix. Luego de leer el libro una persona sin experiencia previa contará con el conocimiento necesario para administrar sistemas Elastix.

Este libro ha sido publicado con una licencia libre porque creemos la importancia el apoyo de la comunidad. Esperamos nos ayuden a encontrar errores, corregirlos y proponer nuevos textos para convertir este libro en la mejor fuente de referencia posible para Elastix. En ese sentido haremos una mención a todos quienes colaboren a mejorar el libro para futuras ediciones.

Desde hoy Elastix es la única distribución basada en Asterisk que cuenta con dos libros escritos que se pueden descargar de manera gratuita desde Internet y la única con un libro en Español. El primero libro escrito para Elastix es “Elastix Without Tears” por Ben Sharif con licencia CC.

Ya lo estuve leyendo un poco y les digo que esta muy bueno el libro sobre todo la introducción donde se tomaron la molestia de introducirnos a la Telefonía con detalles históricos y técnicos, también cabe señalar que a diferencia de  “Elastix Without Tears” este nuevo libro entra en detalles sobre configuración avanzada a lo interno de Elastix, hablando sobre configuración de hardware, del propio linux. etc.

Se los recomiendo.

Emilio Pichardo

Libro de Elastix

En fin, a petición popular de la gente del curro (en realidad es un único individuo… xD), vuelvo a escribir algo en el blog.

Ahora me ha dado por la VoIP.. y es que ha salido un proyecto bastante interesante acerca de las comunicaciones telefónicas a través de las redes de datos (o más bien, cómo transportar una línea análogia a través de unos cuantos enlaces wi-fi) El caso es que ya tocaba pegarse con Asterisk ®, algo que, en mi opinión, está llamado a convertirse (si es que no lo es ya) en uno de los grandes “inventos” de la década, al nivel de la supuesta ‘web x.0′ (hay quien dice que ya vamos por la 4.3 o cosas así). Vamos que tiene futuro la cosa.

Asterisk equivale a una centralita al uso (PBX para los exigentes), sólo que al más puro estilo “pulp fiction” de los servicios TI más conocidos. Para empezar a cacharrear, casi lo mejor es descargarse Elastix o Trix-box, asi podrás echarle un buen ojo a lo que este software es capaz de hacer. En concreto Elastix (que pese a lo que digan por ahí, parec claramente un fork de Trix-box) puede ser la mejor opción por la cantidad de extras que puedes añadir en cuestion de 5 minutos, además parece algo más funcional.

(Sigue leyendo…. !!!)

En refinitiva, Elastix, es un CentOS (creo recordar) con una instalación base de asterisk + add-ons y su front-end ajax pertinente (si, si lo dudabas, esto también usa AJAX) a lo easy-to-use. La ventaja de esto, es que tiene una cantidad ingente de módulos cargados para las tarjetas (Analógicas o RDSI, incluso para primarios) más comunes, y un botonín en la web de gestión para autolocalizar el hardware (qué, flipando ehh?) y es que, esto de la VoIP ya está muy avanzadillo.

Es una buena idea comenzar con algo así, para probar hard y soft rápidamente, y familiarizarse así con el sistema, para luego pasar a personalizarse la instalación de Asterisk en una Debian, aunque, por lo poco que he podido probar, Elastix parece un sistema lo suficientemente estable como para dejar en producción (con cautela, eso si). De hecho, estoy pensando en utilizarlo en casa, con un proveedor VoIP y un número virtual de esos ;) La videollamada es un hecho !!

En términos ecónomicos, no hace falta ni preguntar, sólo hay que irse a lo que ofrece, Cisco, por ejemplo, y echar cuentas para unos 200 usuarios. Entre teléfonos, y la PBX en sí… puff. El ahorro es bestial, y las posibilidades, 1/2 ilimitadas.

No voy a poder hacerme un manualillo, así que no voy a decir aquello de “si tengo tiempo… posteo el manual” aunque supongo que subiré los scripts caseros más útilies (algo es algo).

Por cierto, estoy viendo “how it works” en el discovery, y, o ya estoy que desvario, o creo que podría llegar a fabricar un boomerang con mis propias manos..

Esto es un cachondeo.

Funcionalidades de elastix:
* Soporte de Video, para utilizar videoteléfonos.
* Virtualizacion para correr múltiples instancias de Elastix en el mismo servidor.
* Interface web amigable.
* “Fax to email” .
* Interface para facturación.
* Modulo para campañas de llamadas salientes.
* Configuración gráfica para parámetros de red.
* Reporte de uso de recursos.
* Reinicio y apagado remoto.
* Reporte de llamadas.
* Módulo de voicemail integrado.
* Interface web para voicemail.
* Operator panel integrado.
* Módulos Extra SugarCRM y Calling Card incluídos.
* Interfaz de ayuda embebida.
* Servidor de mensajería instantánea Openfire integrada.
* Soporte multilenguajes.
* Servidor de mail integrado para múltiples dominios.
* Webmail

Con hylafax se pueden enviar y recibir faxes (servidor de fax).

http://www.elastix.org/