[asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend

[Alec Taylor]

Hmm... it's just that I've seen implementation of this done already,
in Google Voice, BlindSide, BigBlueButton and others, however none
provide a simple interface for voice-only broadcast from the browser.

I'm sure there's a way to do it using Asterisk, I just don't know of it!

Please suggest ways of hosting mic-stream-input using a ready-made
project for my website.

Thanks,

Alec Taylor

On Wed, May 4, 2011 at 3:20 AM, Leif Madsen
<leif.madsen at asteriskdocs.org> wrote:
> On 11-04-30 03:10 PM, Alec Taylor wrote:
>> Good Evening,
>>
>> I'm setting up an Internet Radio website with call-in functionality,
>> and need to know the kinds of FOSS tools I should install to get the
>> job done.
>>
>> Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png
>>
>> Call protocol:
>> [Producer calls in]
>> [Host calls in]
>> [Guest calls in]->[Screened by Producer, if accepted, conferenced into host]
>>
>> On the website they'll need to be able to call in (mic input grabbed),
>> and listen in (without calling in).
>>
>> ____
>>
>> I've been suggested many things, including Skype, IceCAST and
>> [currently the most promising] Asterisk+Red5+Red5Phone.
>>
>> Are there any better ways of doing this, and if not, how do I setup
>> asterisk for the above task?
>>
>> Thanks for all suggestions,
>
>
> While it isn't a free option, using chan_skype (license purchased from Digium)
> may be the easiest solution here. A lot of people know what Skype is, and may
> already have it installed, and instead of requiring your users to install a
> softphone, they could just click the "Call Me" link you provide.
>
> Alternatively, you could get a DID and allow people to call in the "old
> fashioned way". You'd have to pay for the interconnectivity, but if you use a
> SIP provider the costs should not be unreasonable (1-2 cents per minute).
>
> You could always look into using Google Talk as well :)
>
> Leif.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>