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 @
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.
If I am correct, it 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 @
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer&view_comment_id=14966:
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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