asterisk users - Jun 2004 - Asterisk as Internet Talk Radio PBX system

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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> 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
> 
> 
This will work fine with a regular (but DUPLEX) soundcard, i.e. most $40-50
soundcards, but based on your mixer, you may need some matchboxes that do
impedance and level matching for you.  The soundcard in most pc's (unless
you spend big $$$ for a pro one, which I recommend against) will have levels
that are too high for your pro gear.  You might get away with just padding
the input and bringing down the gain on the mixer input.

To use Asterisk as platform for such a system you probably want to have a
Alsa enabled card which supports routing of multiple channels in and out.
So Asterisk is like the intermediate 'engine' that routes the signal.
(Or
sort some soft-mixer). A user is then placed in a Meetme room and the
hold signal would be the live show?

OUTPUT					  ||==============o
					  ||
Soundcard 1 (studio/broadcast)           =||=|---\
                                             |    |
Soundcard 2 (prelisten/desk)             ====|\   |
                                             | |  |
SIP/IAX incomming                        ====|/   |
					     |    |
Meetme					 ====|---/
					     |
Meetme					 ====|
					     |
Meetme					 ====|

Technical picks of the phone by prelistening, transfers it to a new or
existing meetme. When the actual 'meeting' starts, the meetme gets in
the
air studio audio (presenter) gets in by the Alsa interface.

Somebody earlier suggested Asterisk for use in remote broadcasts (on a
location for example). With two boxes and a ISDN line on both sides, some
encoding and you are in business too.

Asterisk as a application platform is quite powerfull, but probably has
some overhead which 'all-in-one' products have.


Stefan

Hi!
> 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 
CLI:
show application ices
show application meetme

Also, have a look at Teamspeak.

Cheers, Philipp

James Sutton wrote:
> 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    
I am working on this very thing at the moment, although on a single
box.  Some specifics so far:

Using Icecast/Ices to stream ogg.
The stream is connected to a MeetMe "on-air" conference room.
Music or whatever is connected from the line-in jack on the sound card
using alsa and ices.
Callers can come in VOIP or PSTN and are placed in "hold" conference
room until they are bridged into the "on-air" room.
Callers on hold listen to the live stream by using an application
"OggPlayer" (a modified MP3Player application) that connects ogg123
into their room.

I'm not currently doing any line-out (such as you want to send to your
mixer), but plan to do so.
> I wonder if there is a group discussion of this type of
functionality.

Perhaps we've just started one :)

Barton