similar to: ASTCC/ 'L' option hangup wackyness

Hi I get an error popping up when the call ends as follows: DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at /var/lib/asterisk/agi-bin/astcc.agi line 90, <STDIN> line 32. Does anyone else get this same thing? Looks as if my database table is wrong, or something else is up...not sure Thanks Clive

Hi, I am using ASTCC and trying to setup a calling card platform. The problem that I have is that astcc does not hangup calls correctly: 1. If I try to dial a number, call goes through fine. When I hang up the call from my side I get this: -- Called 192.168.1.56/1XX6872XXXX (masked a few digits) -- SIP/192.168.1.56-086c5000 is making progress passing it to

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

Hi Is anyone having this weird trouble with bristuff....calls don't get answered, but on the debug it looks as if the call is answered. The call will just hear ringing. I am using -BRI-stuffed-0.1.0-RC4a Occassionally the system starts working on its own, other times I have to stop and rmmod everything and start them all up again. These error messages appear on the CLI: == Primary

does anyone had an experience with not hanging up the call in astcc, thats my problem sometimes call does not hang up automatically and even when pressing star the call continue without stopping. Regards; jonny ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive

Hie! I've installed Asterisk CVS-HEAD with ASTCC. The problem i'm facing is that the astcc.agi script completes when the recipient picks up the call. When the astcc.agi completes is returns 0 bill time but both end still able to talk. It occurs intermittently, any one facing the same issue? Asterisk Console ----------------- == Spawn extension (sip, 7777771111, 2) exited non-zero on

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.

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

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

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

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

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

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

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, 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'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

I try to install astcc. Make install gives me: DBI connect('database=astcc;host=localhost','astcc',...) failed: Unknown database 'astcc' at ./astcc-admin.cgi line 67 DBI connect('database=;host=','',...) failed: Access denied for user: 'root@localhost' (Using password: NO) at ./astcc-admin.cgi line 60 Ignoring that (!!! ;-) ) and going to the

Hi, When trying to compile ASTCC i am getting the following error: root@asthome:/usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.6/i486-linux /usr/lib/perl5/5.8.6

I have this error compiling ASTCC: [root@pbx astcc]# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Can't locate DBI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0