this one (written by Daniel) http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb maybe a bit outdated but still consistent ? the thing i am really stuck with (and concerning real-time) is that none of my extensions (from asterisk CLI) are online: ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] that drives asterisk crazy ! and logger reports: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) every time i place a call. The tutorial written by daniel mention a channel configuration pretty minimal: INSERT INTO sipusers (name, defaultuser, host, sippasswd, fromuser, fromdomain, mailbox) VALUES ('102', '102', 'dynamic', '102', '102', 'yoursip.com', '102'); and since there's no context associated to the 102 extension i cant figure out where that channel enter the dialplan ? [public] [LocalSet] [default] ???? and a dialplan exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,101,Voicemail(${EXTEN},b) exten => _1XX,102,Hangup i am sorry to bother you with issues more asterisk oriented than kamailio. By the way i took a good start with kamailio as it seems to work flawlessly on my system. thx you. On Mon, Feb 15, 2016 at 12:26:06PM +0100, Sébastien Brice wrote: Which tutorial? That was a part of a tutorial I once saw. In essence asterisk uses the kamailio database, UA registers on kamailio and is stored there, asterisk sees the same data (realtime). Sébastien BRICE VoIP, Support et Intégration