[RDD] Any one seen an Asterisk module for talk show call in?

Danny danny at aercomm.net
Mon Nov 8 14:42:21 EST 2010


I'm actually in the process of developing just such a system. It's 
sitting on the back burner right now, as some family issues have taken 

The actual hybrid functionality is being done with the Asterisk JACK 
application. You'll want to run JACK at 8khz, as the rate conversion 
code in app_jack.o is quirky. By quirky, I mean that if you don't run 
JACK at 8khz, JACK will crash upon attempting to transfer a call to 
app_jack. At least that was the case with the version that I was last 
working with. It's been a few months since I touched it.

What I'm building is a web-based interface to it all, interfaces for the 
call screener, on-air talent, etc... It's designed from the ground up to 
handle multiple studios and multiple call-in numbers, where any number 
can be routed to any studio. Call screening can be done with a laptop on 
a WIFI connection in the next room using a soft phone or even by someone 
a thousand miles away, so long as they have an internet connection and a 
phone line.

I figure if nothing else, there would be an interest in it among the 
LPFM, podcast and internet radio station crown. Though, my goal is for 
it to be of sufficient quality to market to "real" radio stations.


Bill Putney wrote:
> I know this is a little off the Rivendell topic but where else would i find more Linux/Broadcast folks in one place?
> Does anyone have or heard of a module for the Asterisk Linux VoIP phone system to make it useable as a radio station PBX?
> We're building a new station and I'm wondering if I really need to invest in expensive telephone hybrids. Seems like the wrong way to go at this point in time. We'll be using VoIP for the call in lines and if I have to run them through hybrids to get them into the digital console that just seems wrong! Two more 2 wire to 4 wire hybrids and 2 more A/D-D/A conversions to get a digital phone conversation into a digital broadcast console (sigh). 
> A nice transcoder and an AoIP (LiveWire) module with producers console in Asterisk would be a nice way to go.
> Telos is the only one making a VoIP to AoIP "hybrid" but it's pricey. Everyone else in that market seems to think their 2 wire POTS hybrids are the way to go.
> - Bill 
> Sent from my iPad
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