I fell into voip years ago when Asterisk had just been released, then I working in call center as an IT admin. Since call centers spend all their budget on telecom I read up on Asterisk to see if it could help save some money. I spent some time playing with Asterisk@Home and after getting a feel for the user interface I started digging into the conf files to see how the system worked. Getting into the conf files I was able to come up with my own more efficient contexts which I kept improving upon. Then as business needs dictated new features I would implement, using the Asterisk AMI I added click to call to our landing pages. Then using my knowledge of Asterisk, AMI and the CDR database I wrote an Asterisk call center reporting suite with PHP, AJAX and a MySQL DB.
Since then I have used OpenSIPS, Kamailio, FreeSWITCH, Yate, pretty much any open source voice over ip application to see what I can do with it. My recent focus has been on FreeSWITCH since it is a scalable application and very versatile being able to do almost any aspect of voice over ip, such as a soft phone or a carrier level voice switch. I am really impressed with the group of people working on the FreeSWITCH project and the variety of modules that are available now is really astounding and just keeps getting better.
Just to give you a little more about my background I received a bachelors of science in computer engineering while attending college. During my time in college I started playing with linux and worked for a few companies as an intern doing everything from quality assurance and web development to network administration. The years I spent as an intern gave me the experience needed to get me to where I am today. Training and teaching myself new technologies is something I find is very important to survive in the ever evolving IT field.
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