asterisk users - Nov 2003 - Streaming channels from Asterisk to the Internet

Hi folks,

I'm wondering if it is currently possible to configure Asterisk to
forward the conversations from 2 channels into a streaming daemon,
such as Icecast, so that other people connected to the Internet can
listen.

The concept is similar to a radio talk-show.  The show host would
connect to Asterisk via an X100P or through VOIP.  His or her voice
would then be sent to the streaming daemon for those on the Internet
to hear.  The show host would also have control of the other incoming
channels (via a custom web-interface), which would come in via an
X100P or VOIP as well.  The show host and the chosen channel(s) could
have a conversation streamed out to the Internet until the channel is
disconnected by the host.

Any input regarding the feasability of this, and the available
software (such as asterisk-perl) that can be used to accomplish this
would be greatly appreciated.

Barton

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.

This would be the ideal setup, but if you have only one
computer it is a bit harder. One way to do it would be to
have two audio cards and loop the Asterisk output into the
Icecast input, which is hard to get the audio to go back to
the person on the telephone -- maybe use a conference and
a SIP phone.

Otherwise, maybe Icecast can be hacked a bit or glued to a 
sip client i.e.:

	sipclient sip:12345@localhost | icecast 

for some hypothetical sip client that just listens and sends
audio data to stdout. 12345 would, again, have to be a conference
and there would need to be some other way of joining the 
conference.

As far as managing the incomming call(s) you could use astman
and/or queues...

-w

On Fri, Nov 14, 2003 at 12:18:50PM -0600, Barton Hodges
wrote:
> Hi folks,
> 
> I'm wondering if it is currently possible to configure Asterisk to
> forward the conversations from 2 channels into a streaming daemon,
> such as Icecast, so that other people connected to the Internet can
> listen.
> 
> The concept is similar to a radio talk-show.  The show host would
> connect to Asterisk via an X100P or through VOIP.  His or her voice
> would then be sent to the streaming daemon for those on the Internet
> to hear.  The show host would also have control of the other incoming
> channels (via a custom web-interface), which would come in via an
> X100P or VOIP as well.  The show host and the chosen channel(s) could
> have a conversation streamed out to the Internet until the channel is
> disconnected by the host.
> 
> Any input regarding the feasability of this, and the available
> software (such as asterisk-perl) that can be used to accomplish this
> would be greatly appreciated.
> 
> Barton
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
--

>Otherwise, maybe Icecast can be hacked a bit or glued to a
>sip client i.e.:
>
>         sipclient sip:12345@localhost | icecast
>
>for some hypothetical sip client that just listens and sends
>audio data to stdout.
Fortunately such a SIP client actually exists: "playSIP"; see 
<http://www.live.com/playSIP/>

You can run (e.g.)
         playSIP -a sip:12345@localhost | <whatever>
(the "-a" option means: output the audio stream data to stdout)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>