Hello all, I have a problem with ASTCC. When I create all my routes, I not able to get the destination pattern I desire. I see it come up, but ASTCC seems to select the first available pattern, and not necessarily the exact one I want. I found the MYSQL statement in astcc.agi: SELECT * FROM routes WHERE "$number" RLIKE pattern ORDER BY LENGTH(pattern) DESC; This returns the desired route, but also other routes that may be first in the select, and ASTCC uses that instead fo the exact matching route. How do I get ASTCC to select the routes starting at the begining of $number and not just anything that matches an expression in $number? Thanks for your help. Voipcarib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041214/51a9691d/attachment.htm
Hi,> I have a problem with ASTCC. When I create all my routes, I not able to get > the destination pattern I desire. I see it come up, but ASTCC seems to > select the first available pattern, and not necessarily the exact one I > want. I found the MYSQL statement in astcc.agi: > > SELECT * FROM routes WHERE "$number" RLIKE pattern ORDER BY LENGTH(pattern) > DESC; > > This returns the desired route, but also other routes that may be first in > the select, and ASTCC uses that instead fo the exact matching route. > > How do I get ASTCC to select the routes starting at the begining of $number > and not just anything that matches an expression in $number? >Try with: ^01154.* being 54 the country code for Argentina and 011 the internacional prefix. -- Nicol?s Gudi?o Buenos Aires - Argentina
Bilal Ghayad wrote:>Dear Sebastian; > >Thanks a lot for your kindly advise to use ASTCC. > >But can u advise me the link for ASTCC to download it and wether it is open >source (to download the source and work on it? > >Regards >Bilal > >_ >check it out http://www.voip-info.org/wiki-ASTCC regards>______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >
Dear Sebastian; Thanks a lot for your kindly advise to use ASTCC. But can u advise me the link for ASTCC to download it and wether it is open source (to download the source and work on it? Regards Bilal
Here is the link http://www.voip-info.org/wiki-ASTCC SA -----Mensaje original----- De: Bilal Ghayad [mailto:bghayad@aloonet.com] Enviado el: Martes, 14 de Enero de 2003 18:21 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] ASTCC Dear Sebastian; Thanks a lot for your kindly advise to use ASTCC. But can u advise me the link for ASTCC to download it and wether it is open source (to download the source and work on it? Regards Bilal _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi I installed ASTCC and got it working, when i enter the pin number and dialled the number needed, it says this call will cost point 20 cents per minute, can i get a message like you have 40 minutes and 30 seconds than giving the per min rate ? Thank You Kani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050313/2bf9d9cf/attachment.htm
HI I installed ACTCC, when i enter the pin number it says this call will cost 4.04 cents, it does not give a message like you have 100 mins. how do i get a message about the no of mins i have Tks Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050322/d43ca7c0/attachment.htm
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@odisok.net] > Sent: Thursday, April 21, 2005 1:48 PM > To: junk@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 >
For everyone's information and so it is on the list somewhere, there is a copy of this at http://voip-info.org/tiki-index.php?page=ASTCCGuide This also includes the explanation of the CDR problem.> -----Original Message----- > From: Dave Kettmann > Sent: Friday, April 22, 2005 8:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] ASTCC > > > 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@odisok.net] > > Sent: Thursday, April 21, 2005 1:48 PM > > To: junk@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 > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Is there an easy way to make Astcc silent, so that it does not tell the user how much money he has and the cost to a location, but rather does call control silently. Whats the general consensus of astcc vs areski. Thanks for your advice in advance. Shidan
Look at the source code. It provides for an "option" to be passed from the dial plan to make it silent or give less information. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Shidan Sent: Friday, May 20, 2005 2:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Astcc Is there an easy way to make Astcc silent, so that it does not tell the user how much money he has and the cost to a location, but rather does call control silently. Whats the general consensus of astcc vs areski. Thanks for your advice in advance. Shidan _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes. Any idea how to do? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060327/63a2f34d/attachment.htm