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...;); $query = odbc_exec($conn, "INSERT INTO dialed(number) VALUES('$my_var')"); Even if I only show my_var value or try to use it inside asterisk, the value is NULL. There is another way to do it? Am I doing a mistake here? I'm using Asterisk 1.2.13. Thank you all. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070110/d53d8af3/attachment.htm

...ah host=dynamic username=teste context=from-spo disallow=all allow=g729 allow=ulaw [teste1] type=friend secret=blah host=dynamic username=teste1 context=from-spo disallow=all allow=g729 allow=ulaw EXTENSIONS.CONF exten => 1,1,Dial(SIP/teste1) exten => 0,1,Dial(SIP/teste) Thanks. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061120/8481d7f2/attachment.htm

...but didn't for SAYDIGITS and odbc connection. How can I SAYDIGITS of my_var or insert my_var value into a db? - What I need more to use WAIT FOR DIGIT? Because it didn't stop to wait for digits. - STDIN shoudn't get the result of READ or GET VARIABLE? Where these values go? -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070112/e21de70d/attachment.htm

...y=yes nat=no host=dynamic canreinvite=no context=internal [teste2] type=friend username=teste2 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal2 [teste3] type=friend username=teste3 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal3 -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/185dcf2f/attachment.htm

Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

Hello all! I'm currently using Asterisk in conjunction with a2billing and everything seems to be working great so far. Now, all I'm missing is some sort of a GUI to monitor all calls going out through my trunks. I can always do 'sip show channels' or 'sip debug' from the console but I was wondering if there's anything that basically does the same thing but in a nicer,

I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? -------------- next part -------------- An HTML attachment was