In the last decade, legislation and the advent of Voice over Internet Protocol (VOIP) have revolutionized the communication industry with new technologies, business models, and service providers. Software and commodity hardware now provide an alternative to expensive carrier equipment. One can implement extensible call switching and voice application logic in Open source software applications, such as Asterisk and FreeSwitch. These new application stacks, however, usher in new complexities and challenges, requiring new skill sets to deploy, develop, and maintain. Deploying telephony services requires knowledge of voice networking and codecs, hardware or services to bridge servers to the public phone infrastructure, capital investment in hardware, and ongoing collocation of that hardware. These burdens are a mere prerequisite to developing the actual application, which requires developers to train in new languages, tools, and development environments. Even telephony applications that currently try to leverage a model more similar to web-development such as Voice Extensible Markup Language (VoiceXML), require the dedication to learn a new language and understand telephony interaction. Ongoing operation and maintenance of these services requires teams to adopt new analysis tools, performance metrics, and debugging methodologies. Developing even the simplest of voice services (such as a so-called “phone tree”) requires significant upfront and ongoing investment in specialized infrastructure, skills, and operations. Thus, there is a need in the telephony field to create a new and useful system and method for processing telephony sessions. This invention provides such a new and useful system and method.