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 @ 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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
