[Asterisk-Users] ASTCC

[Dave Kettmann]

Chris,

There is no "official" documentation, but here is what I have found in the control panel.

*BRANDS*
This is where you can setup different cards with different Service fees. I'm not sure what the INC column is for, I usually leave it set at 6. I think it sets 6 seconds to the minimum bill time. Service fees and Servie Fee Days is for like a monthly charge. I havent figured out where the Markup field comes into play yet.

*CARDS*
This is where you make the cards. You can get a list of cards or you can make/add money to any card. You are able to use GET style URLs to make your own interface to this. Just add a card and notice the URL. This could be very helpful if you want to build another interface (I just made direct DB calls)

*TRUNKS*
This is where you setup your Trunks(duh?). You can name the trunk, set it technology, and then relate it to a real trunk/peer name on your asterisk box. If I understand correctly, only SIP and IAX work, but I could be wrong. I know SIP works, I have used it.

*ROUTES*
This is a decent attempt at a LCR script. Here you will setup your costs for different providers. For example, if you want calls to Mexico to me $.10 a minute, then in the Pattern field, put '^01152.*' (without the '') And in the Cost per additional minute, put 1000. (The costs are in 1/100th of a penny. 

There is so much that can be done with the Routes. You can specify more than one trunk so if it is possible to go out a cheap provider for one area, then put that first, if not, then it will try the next one. You are also able to charge a Connect Fee and Include X amount of seconds with that fee.

*CONFIGURE*
This is where you setup your DB connection and some other information. Host, Username, and Password are all related to your DB. Card length is how many digits the card number will be. I think the voiceover always says 12 digits (Not sure).

The "Email New Card Info" did not work for me and I left everything else set to no because I didnt neet it.

*IAX and SIP FRIENDS*
I'm sorry, but I dont know what those are for, I havent found a need for them, but maybe my setup doesnt need them.

*CDRs*
Out-of-the-box CDRs do not work. They are broken. Unless another version was released since I d/led it. I just updated the ASTCC entry on the voip-info.org wiki and the quick-fix is there. http://voip-info.org/tiki-index.php?page=ASTCC

*PROBLEMS*
Besides the CDRs, the only problems I have found with ASTCC is that at the 'one minute warning' ASTCC cuts into the call, announces that you have one minute left, then the call is supposed resume for your last minute. If I had to guess, the RTP stream is broken when this happens because after the warning, neither side can hear the other. Unfortunately, I havent had a chance to find out why the voice traffic stops. If anyone could let me know what they find I would appreciate it. I asked Digium about it, and they wanted to charge me their hourly rate to work on it. 

*OTHER NOTES*
If you look on the voip-info.org, there are a couple neat ideas to use with AstCC. I have a box setup that when you dial out, it asks for the pin number then if the pin is right, it will go thru, if not, it denies you. There is also a way to make it go off of the caller ID so no pin is needed. 

These are just some things that I have found working with AstCC. I am not an expert by any means, but if you understand Perl, then the AGI script should be fairly easy to modify to suit your needs. I hope this 'guide' is helpful to someone. I would like to hear of other people's experience with ASTCC.


Dave Kettmann
NetLogic
314-266-4000

> -----Original Message-----
> From: Chris [mailto:listmail at odisok.net]
> Sent: Thursday, April 21, 2005 1:48 PM
> To: junk at irqx.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: [Asterisk-Users] ASTCC
> 
> 
> Is there any documentation on how to setup the ASTCC?    I've 
> got it working, but I don't quite understand what the web 
> interface is referring to.
> 
> 
> Chris
>