[Asterisk-Users] SIP channel question

[Tor Houghton]

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