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

James Sutton jsutton at neoninc.com
Thu Jun 17 07:32:09 MST 2004


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 ---> |
Mixer  |----> |  Encoding |     | streaming |
{  VOIP Calls}-Ethernet--|                |                       |
|      |   PC         | --> | Server      |
{__________}                 +---------------+  <--- 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.
 
 
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