Hi, I've got a question or two about SIP calling channels. As I understand, there is no facility for Asterisk to make outbound calls as if it were a SIP proxy.
As I understand it, it is not possible to add an extention that simply states "if no match so far, try SIP/<url>" (from what http://www.voip-info.org/wiki-Asterisk+SIP+channels seems to indicate?) Is there a way of passing on such a call, to say a proxy, or should the proxy be the first point of call in such a scenario? I've successfully installed Asterisk on OpenBSD 3.3/x86 (thanks to whoever posted http://www.voip-info.org/tiki-index.php?page=Asterisk+OpenBSD+patch on the wiki), and I am intent on learning more. Regards, Tor _______________________________________________ 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
