Hi, I'm trying to use OpenSips as a registration server for Asterisk (assuming we can get presence working ok). Do I need to setup and use MediaProxy (or similar)? Or is the nathelper stuff good enough? I've made test calls from phones behind NAT to opensips to asterisk and I haven't experienced any one-way audio problems. Also, the phones are not allowed to reinvite, because I need to keep Asterisk in the media path.
-- James _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
