On Oct 25, 2013, at 5:43 PM, Jr Richardson &lt;jmr.richardson(a)gmail.com&gt; wrote: Hi All, Starting a new project, roll your own SBC, not a full SBC, just need some minor functionality. I'm interested in deploying Kamailio as a edge device on a VSP for single entry point for hosted PBX's, Asterisk based. I had some wonderful and informative conversations at Astricon 2013, several folks assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow up to delve more into details. I've been researching configs, topology, modules needed, ect... Most of the examples I'm reading about for this scenario are spreading the registrations across many PBX's without distinction. One concept I'm struggling with is having a specific phone register to a specific PBX. phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101 phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101 phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101 I could add a unique identifier to some part of the registration of each phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What I'm not clear on is when the request comes to kamailio, where would I identify what PBX the phone should register to and how to re-write the 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX and ensure rtp flows through kamailio. Could this function be derived using dbaliases or possibly using dispatcher with group number for each customer PBX? So assuming I can get the registrations to work properly, would standard invite for calling just work or would I also have to have specific config in place to ensure an invite from customer A phone also reaches the correct customer A PBX? A point in the right direction? Thanks. JR -- JR Richardson Engineering for the Masses _______________________________________________ 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