similar to: Calling Card App

Hello, I have a CAC Channel Bank I with FXS cards. I've the system up and running, with just 1 issue. When I make an inbound call, Asterisk says "Zap/26 is ringing", however, the phone never rings. No lights are lit on the CAC during the calll. Outbound call works no problem, and the CAC lights up correctly. Any ideas what could be the problem? -- Richard Cook

Hi, I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ? SW -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am hereby announcing the immediate availablity of ASTCC for *alpha* testing. ASTCC is an AGI script and CGI script which greatly simplifies the task of creating a calling card application on Asterisk. Just check it out of Asterisk CVS as module astcc: export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot cvs login (password is anoncvs) cvs co astcc ASTCC is, of course, distributed

Hi, im trying to setup ASTCC but I'm getting it difficult. I've correctly set up the mysql database astcc and added a brand, trunk, route and a card as follows: brands +------+----------+------+--------------+------+--------+------+------+ | name | language | inc | publishednum | did | markup | days | fee | +------+----------+------+--------------+------+--------+------+------+ | FWD

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

> I use the latest ASTCC version, but I do not see how you can > use different cards / different prices. Can you explain that for me, > please? The cards have a field called "markup", that would mark up the price specified in the routes table by a certain percentage. So it's a card-specific, non-route-specific premium on the base price. Nabeel

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

I try to change something in ASTCC, but I am now totally blind, .... I hang on one line now. I changed: vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi 22c22 < # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN}) --- > # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN}) 35c35 < # exten =>

hello *'s, Astcc not workin what is correct format for defining 1-database 2-brands 3-trunks 4-routes i define all these things but not workin may be i define in wrong format.I have FXO card installed.can anyone implement it and also my sip phone generates very loud noise wat is that i tried several settings but not hear any voice just noise. sip.conf [general] context=from-sip port=5060

Hi, Does anyone have a neat idea as how to bill inbound calls per minute(second) real time? I've been pplaying with astcc, but while 'billseconds' stays empty, 'billcost' has strange behavior - either stays ampty or takes ONCE the "Connect fee"(if I put one) and keeps it that way no matter how long the call is ...( if no "Connect fee" -stays empty). i.e.

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Hi, alhtough i googled for details concerning ASTCC i did not found an aswer to the following: Should i put in my extensions.conf the configuration of the astcc? I ask this because as i see it, in the end of the extensions.conf there is an include statement : #include /var/lib/astcc/astcc-exten.conf Should the config been done in the astcc-exten.conf file or the initial extensions.conf

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

I (cannot sleep and I) am thinking if there is a way to make inwards billing easy possible. To dial out we use something like: exten => _9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF}) (I have an extra field TARIFF, what allows me to use different prices for different users) To dial to a phone we use something like: exten => 888888888,1,Dial(SIP/6001,20,tr)

Hi guys, do you know if it's possible to handle more than 1 call per card with astcc ? Thank you.

I'm trying out ASTCC. I set the card length to 10, and generated a test card. 10 digits. I set the extensions file to: exten => 9175954700,1,Answer exten => 9175954700,2,DeadAGI(astcc.agi) exten => 9175954700,3,Hangup I dial in and the prompt tells me to enter my 12 digit PIN, not 10 digits. How come it thinks it is 12 digits? I set both the Published number and DID in the Brand

Hello, I'm trying to setup an asterisk extension to be attached to an H.323 gatekeeper so that an asterisk application (Astcc) answers H.323 calls from any terminal logged into the gatekeeper. I'm using asterisk's channels/h323 implementation, and I've configured the following in h323.conf: [general] port = 1720 bindaddr = AAA.BBB.CCC.DDD allow=all gatekeeper=XXX.YYY.ZZZ.AAA

Hi list. I?m using ASTCC with callerid authentication and got things working fine, except for one single issue: Using this command -- DeadAGI(astcc.agi|${CALLERIDNUM}|${EXTEN:1}|5) -- and passing the 5 parameter for silent, it exits unexpectedly. I tested the 4, 3, 2 and 1 and they are working ok. But... when I use the five it returns zero and exit without executing the dial command! Take a

I am new to the list, so if this question is redundant, please point me in the right direction for reading. I want to setup some calling cards for fundraising. I have ASTCC installed and working, but I am wondering how things might work once in production. A customer calls an 800 number (sixTel) and then dials Mexico (voipjet). This all stays IP and doesn't tie up phone lines. I've

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've