Thanks for the help! There's a doc called "Kamailio-Start-To-Finish.pdf", that is super simple. I've got a phone registered to Kamailio, and can call a phone on asterisk just fine. Now when I call Kamailio phone from asterisk phone, the call just loops back to my asterisk box. I'm sure I'm missing something since this is the only dialplan logic I've currently configured. route(TOASTERISK); } # Route ToAsterisk route[TOASTERISK] { rewritehostport("X.X.X.X:5060"); t_relay(); exit; } I think I need a FromAsterisk route; just need to figure out where.. Matt Scott From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Stoyan Mihaylov Sent: Thursday, November 15, 2012 6:19 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio getting started My configuration is SIP client <-> Kamailio <-> Asterisk <-> Kamailio <-> SIP client You can start from http://www.kamailio.org/wiki/start#tutorials Dialing from Asterisk looks like: SIP/KamailioIP/NumberToDial,gwWL(3307996) In Kamailio if you put IP of Asterisk servers in correct tables, you can authenticate calls with allow_source_address() WIth ds_select_dst("1","4"); you can forward call to Asterisk servers. In Asterisk, you should allow calls from Kamailio.... On Wed, Nov 14, 2012 at 8:10 PM, Scott, Matt <mscott@homeadvisor.com<mailto:mscott@homeadvisor.com>> wrote: Anybody got a good tutorial or starter project for Kamailio? We have Siremis 3 installed, but there is 0 help on that? Trying to solve 2 scenarios: -asterisk to softphone through kamailio -softphone to asterisk through kamailio. I ultimately want to use Kamailio as an edge router Any help is appreciated. Matt Scott _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users