Hi everyone I followed this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and got it working (101, 102 and 103) can call eachother. But now i am trying to figure Asterisk's role out. I am more an ipbx person and i am used to register providers trunk in asterisk/sip.conf file, like this: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension] doing that i got plenty of OPTIONS request and 200 OK reply between my Kamailio and my provider (and it is a bit noisy) doing that i feel missing kamailio's logic and power to deal with externals trunk provider The thing is i need my authenticated users (101,102,103) be capable dialing my trunk and requesting INVITE for non-local request. What is the best way to achieve that? My DID provider gave me user/passwd/realm. I heard about avp special variables (auth_XXXX_avp and uac) and some snippets config that could help me to go there. Is that efficient to place the routing's logic to Kamailio and how to do that with my ovh trunk? thx you _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users