[Asterisk-Users] FXO registration and VegaStream

[Mike Fedyk]

Peter Doyle wrote:
> I figured asterisk was looking for SIP user 06, so I added it, but I
> still got 404's.  Turns out I just needed an EXTENSION, 06.  I can now
> make calls and receive them, too.  Of course, if you have multiple
> incoming lines, you'd need extension 06, 07, 08 ... etc, since each port
> has its own "Interface Number" (by default), to allow routing of calls
> made to different lines.
Yes, that's right.

You should specify a separate context for the incoming lines and if your 
port numbers relate to extensions do something like this:

exten => s,1,dial(sip/20${EXTEN},,o)

That way when you dial port 01 on the vegastream, it will ring on the 
sip extension 2001.

This page explains the s context more:

http://www.voip-info.org/wiki/index.php?page=Asterisk+s+extension