[Asterisk-Users] How do I make Asterisk login at/use VoIP provider?

[BK [address only for mailing lists]]

Hi

please excuse if this seems obvious, but I am new to this and the SIP 
section in the Asterisk handbook do not give any clues nor do the SIP 
examples in there seem to represent real-world situations.

I am using Nikotel as a VoIP provider (for now) and I would like to 
configure Asterisk to sign on with Nikotel so that I can use the 
telephones connected to Asterisk to make calls using the Nikotel service.

Checking the preferences in Nikotel's softphone to get a clue for what 
the settings are, here is what I found:

Server: calamar0.nikotel.com
Service: nikotel.com

I assume that I have to use one of these in sip.conf with the register 
directive, but which one?

Further, if my username at Nikotel is "fred", where do I specify this? 
Do I specify

register => fred at calamar0.nikotel.com

or

register => fred at nikotel.com ?

and where do I specify the password? Do I do

register => fred:blah at nikotel.com (as one would do with a web browser)

Also, I don't quite understand the /nnnn at the end of the register 
directive. Do I *have* to specify this? And if I do, what does it do? If 
I have

register => fred:blah at nikotel.com/1111

does that mean that only extension 1111 can use the Nikotel service?

If so, do I have to specify multiple lines with register, one for each 
extension or can I just omit the /nnnn ?

Further still, the examples in the handbook only show "friend" but not 
"peer",

I assume I have to define Nikotel as a peer. How would I do this? Do I do

; [Nikotel]
type = peer
username = fred
secret = blah
host = nikotel.com

or

host = calamar0.nikotel.com ?

and how does this relate to the above "register" business, why specify 
this twice?

Finally, how do I dial? At present, if I dial 9, it goes straight out on 
channel Zap1-1 (FXO card). What is the best practise for alternative 
routes? Should I have it dial via Nikotel if one dials 8 instead of 9 
for an outside line?

Again, if I wanted to do this, I can't quite see from the examples and 
the explanation in the handbook how I would do this. I guess the example 
on page 37 is in principle the way how to tackle this, but there is no 
example for SIP providers to be used as an "outside line".

I guess something like

[voip]
ignorepat => 8
exten => _8[1-9]XXXXXXXXXXX,1,Dial (???)
exten => _8[1-9]XXXXXXXXXXX,1,Congestion

is in principle how to do this, but how does one dial out using a VoIP 
service as defined in sip.conf ?

Then, is there anything else that needs to be done for VoIP ? Note, my 
Asterisk is behind NAT, but I am not getting any incoming calls from 
Nikotel (at least I don't want them as they're prank calls from online 
voice chat lurkers) so I assume that I don't have to do any port 
forwarding (it works without that using the softphone).


Again, my apologies if these questions seem rather stupid. I'd 
appreciate any help. Thanks a lot in advance.

rgds
bk