similar to: ASTCC speaks and cut RTP channel

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 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

I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by root@............... on a i686 running Linux. I just spent some time in testing this. I tested the local and IAX2 trunks. Both worked flawlessly. Any comments? Darren Wiebe darren@aleph-com.net

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

Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

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

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Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

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

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Hi list! Do anybody has success histories about using the Beronet app_saynumber application? For those that are hearing about for the first time, it?s a piece of software that enables the use of another language in say_number commands in asterisk dialplan or AGI scripts. Link to download: http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz I?m trying to compile it in

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

I have got my ASTCC up and working , but facing a hard time understanding the terms in ASTCC what actually they are meant for. 1. What is Trunk in ASTCC and what are the meaning of the parameters such as Technology & Peer/Trunk ? 2. In the routes if I use 0 connect fee that means the user won't be charged for 0 sec calls right ? Inc. Seconds refer to interval ? If I set 500 in cost per

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 list! I?m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I can?t understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: "Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742

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

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

I am attempting a fresh install of ASTCC on Ubuntu. Getting install invalid user as bellow. Has any one seen this? Can some one help out? /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 Detected dry run! ./astcc-admin.cgi >/dev/null DBI