Re: [asterisk-users] asterisk on debian
On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? Depends on the version you want to install. You can install with apt-get install asterisk, of course. More info here: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian Regards, Victor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
Or, you can just transfer the calls into the conference room. On 3/19/07, Lee Jenkins [EMAIL PROTECTED] wrote: Yehavi Bourvine +972-8-9489444 wrote: Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [G.729] Input Gain
Hello, I have recently buyed the g.729 license from the Digium site and have the following issue: I have two voip providers (SIP). The first uses g.726 and the second g.729. The problem is that the input gain from the second provider is a little lower than the first one. I usually use both, so configuring the input gain on the IP-Phones is not a solution. Can I configure the input gain of a context on sip.conf or maybe can I configure the input gain of g.729 encoded streams. In zapata.conf (or misdn.conf) I know there is the above option (rxgain and txgain). Thank you in advance, Victor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
