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
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