[Asterisk-Users] Asterisk as Internet Talk Radio PBX system

[James Sutton]

I see in the archives a brief thread between Barton and w last November 2003 about streaming to the Internet.   I’d like to use an Asterisk to mediate multiple VOIP calls originated from the Internet to the studio to be mixed then passed out to an encoding PC thence back to Internet   {~~~~~~~~~}                 +---------------+                      +-----------+     +---------------+    +---------------+ {   Internet    }                  |  Asterisk  |  --- line out -à |   Mixer  |--à |  Encoding |     | streaming | {  VOIP Calls}—Ethernet--|                |                       |             |      |   PC         | à | Server      | {__________}                 +---------------+  ß- Line in --- +------------+     +---------------+    +---------------+         |                                                                       |    |                                              |         |                                                        ??           cd  |                                              |    Talk Show Callers                                                       Mic                                      Internet  Via VOIP   Notice there need not be ANY telco POTS lines.     I wonder if there is a group discussion of this type of functionality.   Would the LINE OUT/IN   from Asterisk to  analog MIXER console  be PC Sound cards or something more discrete like a form of telco line cards? We do not need the additional freq crunching done, typically, to interface to limited bandwidth telco network.. Jim Wireless Tech Radio www.wirelesstechradio.com I have thought about doing this as well, for what may be the same application. The easiest way to do it would be to use the Console channel and audio drivers and use a mixer -- keep in mind, I'm thinking of a radio talk show, presumably with a mixer, other audio sources, etc. It would look something like this:          +----------+--- line out -->+-------+    +------------------+ POTS --| Asterisk |                | mixer |--->| streaming server |        +----------+<-- line in ----+-------+    +------------------+             |                        | | |               |             |                       CD | |               |     SIP Clients, Etc.                Mic |           Internet                                        Etc.   Where line out of the Asterisk goes to an input of the mixer and line in is connected to a monitor port on the mixer. This would be very simple to do and wouldn't require conferences. You could map inbound calls to some telephone if you wanted to screen callers or anything like that and then forward the call to the console extension when you are ready to go on the air.