Add a PulseAudio channel to Asterisk

Asterisk is a multi-threaded telephony server. It already has channels for the JACK and ALSA sound systems. However, many Linux systems only come with Pulseaudio. Jack is difficult to install+configure, and ALSA frequently doesn't work correctly. Your job is to write a native Pulseaudio channel so that the Asterisk dialplan can call Dial("PULSE/Joe"). [Joe] shall be a section in /etc/asterisk/[login to view URL] describing the address of the PulseAudio server to connect to, which devices (and at which volume) to connect the sink+source to, whether the connection is bidirectional/monitor/announcement, whether in-band DTMF is recognized, etc. The code shall include a "pulse XXX" command for the Asterisk command line which is able to show active channels and their state (including Pulseaudio status such as latency, if online). It shall also be possible to reload the configuration file. (I.e. "pulse show channels", "pulse show channel Joe", "pulse reload", etc., as per established Asterisk conventions) Asterisk runtime configuration, both static and dynamic, shall be supported. The code shall be submitted to Digium for inclusion in the next Asterisk release. It must conform to Asterisk's coding standards. You will need to communicate with them during the project to ensure that your code is acceptable. Your bid should include some information about past projects, a cost estimate and some idea which milestones you would suggest for progressive payment. The past projects shall be at least somewhat related to this project; bids which mention PHP or Android projects will be rejected. Successful submission to the main Asterisk code base shall be one of the milestones.