similar to: simple perl-agi - where's the error?

I have not used astcc with pin codes so far, since I set-up the phone number as card number. Some of my users want now to dial in to the system and than use their card, which is their phone number. For that I would need a way of authentication, like a pin. I want to use something like: What is your card number: <user keys in the number> Enter your pin: <user enter a long pin>

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I've been using for a long time asterisk-perl-0.08 for prepaid card applications, and I've identified a problem with the last releases of asterisk-1.2, installed with Trixbox. The command get_variable() raises a signal SIGPIPE when it is called (whatever the variable to get). I made tests with Asterisk 1.2.20, 1.2.21 and 1.2.22, and I

Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the

Hi, I have set-up astcc with outgoing sip channel. Call processing works fine but after the call tables, CDR and Cards does not get updated. At the beginning it goes to the database and fetch card details and correctly provides the card balance etc. Also it indeed write the inuse field (so writing and reading from database works fine). I've inserted a break point as such in the code;

Looking at the code, it would appear that the 'callstart' column of the cdrs table should really be called 'callend': $dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)"; $res = $AGI->exec("DIAL $dialstr"); $answeredtime =

Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of "hangup". I suppose that astcc.agi was not able to get the answeredime variable from the SIP channel... I have added a few functions to

Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use

hello! has this made it into 1.2.3 already: http://bugs.digium.com/view.php?id=6128 ? i'm trying to set a variable that should be used as a dialstring in the dial-command, including parameters seperated with the respective delimiter, e.g. like: exten => 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh) exten => 907,n,Set(DIALSTRING=${DESTINATION1}) exten =>

Does anyone have a mysql query that will compare a number from the asterisk cdr to a table of international country+city codes to determine the closest match? The two fields are; 1. Asterisk mysql cdr 'dst' field - sample record value '011441316551212' 2. rate table data like this DialPattern 011447977 011447979 011447980 011447981 011447984 011447985 011447986

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

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 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. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

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

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 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

Dialing out If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. If you want your application to initiate a call out without being started through the dialplan: * Asterisk auto-dial

Dialing out If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. If you want your application to initiate a call out without being started through the dialplan: * Asterisk auto-dial

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)