Hi All, I am using Kamilio 3.0.2 as a load balancer in front of Asterisk servers, using Dispatcher/PDT and such, working fine. I would like to be able to bring sip calls into Asterisk at different entry points in the dialp plan, so I want to setup sip users; [sipentry1] contex=blah, [sipentry2] context=blahblah.
In Kamailio, how would I go about receiving a sip request, append a user "sipentry1" then forward it to Asterisk? I would be using some sort of trunk prefix to identify which sip request to append the user like: 552145551...@siprouter, strip 55, append user "sipentry1", dispatch t-relay to asterisk 562145551...@siprouter, strip 56, append user "sipentry2", dispatch t-relay to asterisk 572145551...@siprouter, strip 57, append user "sipentry3", dispatch t-relay to asterisk Any point in the right direction will be appriciated. Which module should I be looking into? Thanks. JR -- JR Richardson Engineering for the Masses _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
