similar to: Example: calling card using extension logic ONLY!

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

Hello I got the similar error while trying a call. -- Executing Answer("SIP/8000104-86ef", "") in new stack -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php:

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

Excellent work! Thanks a lot ------------------------------------- Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use

I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a "beep" I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? Tx, Bogdan

Dear All, Here the version 2.2 a new version of your dear CallingCard Software !!! http://www.areski.net/areskicc-doc-v2/ Many new features have been added and several enhancements made! Newest features : - A new re-build rate-engine - LCR & LCD management (OOOOHHH YESSSSS) - Billing Increment - Progressive Rate - Scheduled Rates (days of the weeks) - Expiration rates - Buy rates

hi all, I have a little problem. I'm trying to configure a2billing (asterisk2billing) with asterisk. Everything done successfully but when I try to call following error occur "WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application 'DeadAGI,a2billing.php' for extension (a2billing, 456,3) and it hang ups the call. Can someone please tell me why this error occuring. My

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You

Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as

I have this situation: My machine debina, has three interfaces: eth1 212.126.24.129 ppp0 10.2.0.1 (Point-to-Point: 212.31.242.98) nsc5 10.2.0.250 (Point-to-Point: 172.23.140.32) I want packets which come in through the nsc5 interface, to be FORWARDED to the ppp0 interface to 212.31.242.98, even when their destination address is 212.126.24.129 (even so this is the IP of eth1 on this

* Read the config sample files! (even if you're an Asterisk guru) ----------------------------------------------------------------- For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS. As we add or change features in Asterisk, the sample

I appreciate any recomendation of a simple CallingCard Application and resources of users manual. __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You

I want to build my own calling card system on Asterisk. I looked at this page - http://www.voipinfo.org/wiki/view/CallingCard+Applications and it has listed some applications that I thought could help speed up the development process though the link down the bottom doesn't work. Does anyone know of any AGI etc applications to build a Calling Card system on Asterisk? Michael

-- Executing unixODBCput("SIP/10-cc1b", "BLAH/blah=bkw") in new stack -- unixodbcput: family=BLAH, key=blah, value=bkw -- Executing unixODBCput("SIP/10-cc1b", "BLAH/blah=bk2") in new stack -- unixodbcput: family=BLAH, key=blah, value=bk2 -- Executing unixODBCget("SIP/10-cc1b", "testingget=BLAH/blah") in new stack -- unixodbcget:

I've had a dedicated box running for ages in my LAN without any kind of problems. Ssh has been installed and useable till tomorrow when a problem pop up. KERNEL: 2.6.5 no server or client settings have been changed. I can ping and nmap the host without any kind of problems. Bellow I'll paste a verbosed ssh try. bkw at tellus ~ $ ssh -vvv neptune OpenSSH_3.8p1, SSH protocols 1.5/2.0,

https://bugzilla.netfilter.org/show_bug.cgi?id=983 Bug ID: 983 Summary: mnl_socket_recvfrom hangs in example code Product: nftables Version: unspecified Hardware: x86_64 OS: Fedora Status: NEW Severity: normal Priority: P5 Component: nft Assignee: pablo at netfilter.org

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:

Hi Hekuran I have installed asterisk-1.0-RC2.tar.gz, asterisk-prepaid-0.3.1 and postgresql. When I tried to call from any IAX client to another IAX client and also sip client to sip client it worked fine. And also the cdr table filled properly. Now I tried to configure asterisk-prepaid-0.3.1 with asterisk. I have compiled asterisk-prepaid-0.3.1 and also copy the configure file. I

Hi everybody, I'm running an asterisk box -cvs version since few monthes, updated it middle of may and a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab