On 3/7/06, Lee A. Azzarello <lee@xxxxxxxxxxxxxxxx> wrote: > Even though I have done it before, I still have the belief that installing > Asterisk locally is overkill for a single person to make calls. In wanting to bridge the world of Linux audio and asterisk telephony I had desires far beyond making a single phone call. I would argue, first, that a local setup of "single user" asterisk could be made a lot easier (freeswitch is veering in that direction) and the flexibility of having a pbx on your laptop or wherever (local voicemail. call presence information. Text-to-speech support. Etc) In particular I wanted to make it easier to do a call in radio station (see rivendell) and integrating the voice to mp3 function and music to mp3 function struck me as asterisk with jack as a natural bridge... If ladspa plugins could be run through asterisk or a jack compliant sip phone you could give your outgoing voice calls a little bass boost for that "voice of god" effect... surround sound conferencing becomes feasible. stuff like that. It doesn't make sense to me that these two worlds - telephony and professional audio - should be seperated. I haven't had much spare time recently to work on these ideas, but freeswitch seems to be a bit more hackable than asterisk has become, so I've been looking at that... > I'm not even > sure how an Asterisk jack channel would function for RTP input to Asterisk. > What would do the signalling? Mentally to me, a jack port is a inband telephone connection, no real signalling save perhaps silence suppression need be used... DTMF, etc, generated in band... -- Mike Taht PostCards From the Bleeding Edge http://the-edge.blogspot.com
