All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 & 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten => _9.,1,DIAL(SIP/${EXTEN:[email protected]).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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