For SIP in sip.conf use the option canreinvite=no for a specifed device. For IAX2 I think you have to use notransfer=yes
Ivan > I would like to force RTP traffic for SIP to go through PBX. Is it > possible to somehow force it in configuration? Is there also possible > for IAX? _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
