similar to: Working with an ongoing call

Hello There, I *think* i've setuped the AreskiCC2 Calling Card system right , but i've yet to make any calls out of it , i added a rate card , trunk and defined some rates , generated some users , added 10 dollars in them , okay , now i call any number , it asks me to enter my pin , i do , it tells me i have ten $ , right after that it says sorry you dont have enough funds for this call

Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald

Hi Folks, After going to the paifull steps of installing AreskiCC and finally being able to access the webinterface, connecting to *, importing rates and setting up accounts I am not being able to make a CALL: No matter what number i try to dial I get the same response: The number you have dialed is currently unavailabel. Please enter thenumber you want to dial starting with 1 for local and

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

Hi All, i have created one prepaid PERL AGI script to integrate asterisk users in our current Oracle Billing System. I am using $AGI->exec('Dial', $dialstr); in script after getting the MAX time out for the priticular call. But when the channels increase on my asterisk more than 50-60 asterisk get crashed and i am suspecting the cause is of AGI Script. because when i check ps on

Hi, I'm interested in anybody that is providing a phone support service using an Asterisk system, with built in charging system. I run a PC support company and use Asterisk at the home/office. I would like to be able to provide technical support to my customers using asterisk. However I want to be able to charge them fairly for this support, and with little work on my part. My idea was for

Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard]

I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and

if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use

I'm looking for a prepaid calling card platform that: * easily scales to multiple servers with a common database for: redundancy, capacity, and performance Looking to start with capacity to handle 100 simultaneous calls and be able to easily scale to 1000+ simultaneous calls. * in addition to the normal anti-fraud measures, supports an API for easily adding new anti-fraud tests along the

First posted at: http://deancollinsblog.blogspot.com/2009/02/amazon-flexible-payment-syst em.html Amazon have just announced they are finally opening up their API's to their credit card processing platform https://payments-sandbox.amazon.com/sdui/sdui/business?sn=devfps/o So does this mean that micropayments have finally been cracked? It certainly helps out users because you can

Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable "call forward". The result of CDR seems not correct. UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number. I think we shall charge the credit from UA 9999 not UA 1011 because UA 1011 don't know where UA 9999 forwards to. But in CDR, I can only find the from(1011) and

Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker "SIP peer authentication on an external database (RADIUS - LDAP)", etc. Although, none of these seems to give me the

I had CC readers going over the internet (with pings over 80ms) connected to Linksys PAP2. It was only successful once every 3 attempts. I had 100% reliability when it was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com

http://deancollinsblog.blogspot.com/2007/03/microsoft-buys-tellme.html I thought I would email this post I made on my blog from yesterday as a way of stimulating discussion on this. It looks like the Asterisk community is no closer to getting a Pre-Paid 'Offboard Speech Recognition Processing' SIP gateway than ever. I've left the

Hi there, Regarding the calling card system, I would like to know what software tools are suitable to write the CDR system? 1) Perl (but what kind of database should be used?) 2) PHP + MySQL 3) or others suggestion? Also, which one is easier to access AGI? Any source and information would be appreciated. Best regards, Max Chow -------------- next part --------------

I am using Asterisk with Connect.VoicePulse. Of late, we are getting too many congestion errors. Chris Icide has helped me before in setting up the server. He has done a wonderful job. It has worked very well until about 2 months ago. Now I need some help to fix this issue. I appreciate the help. Sincerely, Naren Koka (480) 829-0479

Dear list I have been at it for a few day trying to get my SuSE 8.2 server (FREESIDE) to produce a domain (SKOGFARET) for my Windows 2000 Server machine (BABYLON). I use OpenLDAP, and the setup seems to work in most ways, except the vital one [ie. I can mount shares, print, etc...] When I try to add W2K (BABYLON) to the domain, I get the following error: "The following error occured