[asterisk-users] sip.conf & realtime

[hugolivude]

Hi -

I'm looking into realtime and I'm having a bit of a problem with the SIP
part.

My review of the posts seems to indicate that I should use realtime static
for the [general] part of my sip.conf including the registration commands:

register=><did>:<secret>@<domain>/<did context>

and use realtime realtime (funny name!) for peers and friends:

[myprovider]
type=peer
auth=md5
username=...
fromuser=...
fromdomain=...
secret=...
host=...
port=5060
nat=yes
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
context=incoming-sip

Is this correct?  What's throwing me off is this statment found
here:<http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static>
*
NOTE:* You can only store a static config OR a RealTime config. You cannot,
for example, store sip.conf and use sipfriends via RealTime.

This would suggest that I'll have to do a reload when I add a DiD, but a
reload won't be necessary if a new SIP client is added.  Do I have it right?

Also, what's the difference between a peer and a user?  I used to think that
a "user" was an agent  authorized to call in to my * box, a "peer" was an
agent I could reach and a "freind" was both.  What's throwing me off now is
the statement found
here:<http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static>

With newer versions of Asterisk the concept of SIP 'users' will be phased
out.

I can't understand this especially in the context of extconfig.conf that
uses both a sipuser and sippeer entry.  Could someone clarify for me?

Thanks,
H
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