Hi there, I am new to Linux and Asterisk. I am not new to computer, network and telecom stuff but only did 'Redmond-issues' and classic PBXs from Siemens and Agfeo until now. So it took me some days to get linux (debian) and asterisk up and running.
I made it to configure some SIP Extensions, with the help of AMP, and I use X-Ten Lite SIP Softphones to do calls. The Login procedure works fine. I have three internal phone numbers (200,201,202), but I can not make internal calls from 200 to 201 or anything. When calling an internal number X-Ten Lite tells me "404 Not Found" I do not have configured any provider for the outside world. So I do not have PSTN lines or SIP providers like sipgate configured. Now I have these questions: - Do I need to have these outside world "trunks" configured to tell asterisk that when I dial "0" I want to dial outside, and when not, that I want to dial an internal number? - Is it the G729 codec and license problem? - Any other hints? For SIP I also have these questions: - Is an sipgate account busy when I am already taking or making an call? Or can someone else use a "2nd line" of an sipgate account? - Is it possible to have dial-through numbers through an sip-account, meaning my sip-number is for example 123456, then the person with the internal extension 200 will have as an direct call through number 123456-200 ?? Thanks in advance, bye, Michael. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
