VoIP Mailing List Archives

[flavio at opensips.org]

The OpenSIPS project is proud to announce its next Bootcamp.

Join the core developers and founder of the OpenSIPS project for a week of training in the US! This exclusive opportunity is limited to 20 students and represents the only such event for this year. This year's curriculum has been updated to reflect the many innovations introduced in OpenSIPS 2.2, a long term supported release that guarantees knowledge gained has a longer half-life.


Certified training saves time and money

OpenSIPS mistakes are easily avoided if you get proper training! Companies that use OpenSIPS waste time and money when they don't have a trained engineer on staff. Searching google, waiting on IRC, even the latency in mailing list replies takes it's toll over time. Take this rare opportunity to train your employees with the project members themselves.


New updated track for 2017

With new content for OpenSIPS Version 2.2 LTS, this training delivers an increased depth of knowledge on scripting and routing leveraging some of the newest (as well as integrating with some of the oldest and most reliable) tools in the VoIP ecosystem. Our goal is to provide our students with the latest, most valuable, and pertinent knowledge that they can take back to their workplace and begin improving their networks.
We strive to deliver a training environment that is easily replicated and found in most VoiP deployments. In terms of technical setup, students will participate in labs deployed in their own private space as well as interact with other students and theirs...Read more


Key Objectives
• Examine OpenSIPS use case scenarios, architecture, advantages and even limitations. • Install, configure and run OpenSIPS. • Interpret and understand the scripting language and core configuration of OpenSIPS. • Understand SIP transactions, SIP dialogs and NAT handling when it comes to deploying OpenSIPS. • Integrate with a relational database such as MySQL. • Install, configure and use the OpenSIPS Control Panel web interface. • Integrate with PSTN gateways and wholesale providers. • Implement advanced SIP call flows such as Call Forward, Call Hold and Call Transfer.
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