similar to: ASTCC astcc-config.conf card length question

I have been trying to make Broadvoice match incoming Caller ID and do specific things based on the number received, but due to Broadvoice requiring the "s" to start off an incoming extension, I cannot get this to work. Has anyone been able to do this? Here are some examples of my setup: from sip.conf: context=broadvoice-incoming from extensions.conf [broadvoice-incoming] exten =>

I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week?

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 Today I?ve installed, apache 2.0.52, mysql-4.1.7, asterisk-perl-0.08 and ASTCC prepaid card aplication from CVS, so now I have access to the astcc-admin.cgi from web server http://asterisk/cgi-bin/astcc-admin/astcc-admin.cgi and I?ve been able to create the database from "Configure" menu but I have some doubts to continue: - Do I have to reinstall asterisk with mysql support? -

How can I fulfill that? *Billing Increments* Continental USA: six (6) second increments. International: thirty (30) seconds minimum and six (6) seconds thereafter. Mexico: sixty (60) seconds minimum and six (6) seconds thereafter. bye Ronald Wiplinger

What has been changed at the ASTCC from the previous head to the current one? How to use the PIN? Can I avoid it? bye Ronald

Hello, This is debug output I get: Jul 28 15:05:49 WARNING[8249]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_zap.so] => (Zapata Telephony w/PRI) Jul 28 15:05:49 WARNING[8249]: chan_zap.c:924 zt_open: Unable to specify channel 1: No such device or address Jul 28 15:05:49 ERROR[8249]: chan_zap.c:6460 mkintf: Unable to open

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

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: 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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5870 bytes Desc: image002.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050129/e53b0567/attachment.jpeg

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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5870 bytes Desc: image001.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050128/198114e6/attachment.jpeg

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

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