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roll-over

Posted by gregshap on Fri, 08/19/2011

I am trying to get an live talk show system to perform like the old phone company roll-over.

I need the VOIP provider lines to ring a 6 line VOIP radio show system without ever answering the calls (such as a huntlist would do). A hunt list would not work as that actually answers the call and starts the billing minutes from the provider. I just want the inbound trunk to ring the first extension then as another call comes in, it would ring the second extension and so on...through to the 6th extension. At that point, any more calls would be busy and not route.

These calls could be ringing for as long as 5 - 10 minutes until the host takes their call on the air.

Thanks for any suggestions you can come up with !

Greg

MTE Asterisk 1.6


Submitted by eeman on Fri, 08/19/2011 Permalink

it is illegal to do what you are asking for.. the legal amount of time a call is allowed to be in the ALERTING or PROCEEDING state is 120 seconds. This is set on the whole PSTN backbone to avoid the BlackBox fraud that existed in the 80s (if you dont know how a blackbox works I'd be happy to explain). If they are telling you otherwise, then they just don't truly understand how that works. I have never seen a radio station that when you call in you hear the ringing sound for 5 min. Your outbound carrier wouldn't allow you to stay on that long. Try it with your cell phone if you don't believe me :-), either they answer the call and your cell phone starts its timer of call duration (even if you hear fake ringing), or your carrier will tear down the call.

Every radio station I have called into I am answered and put on hold by the sound engineer until the DJ takes my call.

Now if the call can be answered, they pay for the duration time, and they want to play fake ringing to the customer you could do that with a queue using the 'r' option to replace MOH with a ringing sound.

Submitted by gregshap on Fri, 08/19/2011 Permalink

Erik,

I am very aware of Blackboxes, Blueboxes and Silver boxes!

What could I do to at least let it ring for say 119 seconds then have it roll over to an available line?
This would minimize the "Per Minute" inbound DID routing charge for just listening to a ring tone.

They have a hunt list now that eats up $20 - $30 a day from people waiting to 'maybe' join into the talk show. I have tried "unlimited" inbound providers plans but the popularity of the morning show host chokes the providers system to a point of failure. Per minute seems to be the only way to limit the paths to my system.

any solutions would be appreciated,

(Phone Phreak) Greg

Submitted by eeman on Sat, 08/20/2011 Permalink

Im going to make a few assumptions.. like you are feeding the radio station booth with analog service off an Adtran 908 or something similar.

- Point the DID directly to your huntlist so that other scripts do not execute an Answer on the call.
- create several "Device Based Ring Group" in your hunt list
- for each one assign a different channel on your Adtran that corresponds to lines 1 - 6
- for each one give the timeout period 120 and make sure 'i' is in the options
- for your final step, where it says "Destination on no-answer" execute a custom script that hangs up the call.

for your custom script it's a simple 2 liner, call it something like 'return-busy'
exten => s,1,NoOp(Sending Busy signal to carrier)
exten => s,n,Hangup(17)

This will work better than playing a Busy application since Busy will keep the caller connected to your system while they listen to the busy tones, a Hangup(17) will send the applicable channel message to indicate busy (in sip its 486 "Busy Here") to your provider, who then passes that signal back off to the original caller. The original caller's PBX will then play the busy tones locally avoiding that long connection time.

I forgot to mention that on the Adtran or whatever it is you need to disable call waiting so the call count is limited to 1 per channel/line

Submitted by gregshap on Tue, 08/23/2011 Permalink

Thanks Erik,

this seems to have done the trick.

The only difference here is that they don't have an analog device but a new VOIP on-air phone system that takes lines directly in by registering as SIP. It is called a phonebox if you want to look it up at:
http://www.bionics.co.uk/broadcastsoftware/phonebox/phoneboxsolo.aspx

I still followed your suggestion and it worked great.

thanks again

Greg