I use a 'virtual' PSTN line (voip trunk) from (http://www.voxbone.com) as incoming external line to my Asterisk server (192.168.1.100) In my router I have have : Application Start End Protocol IP Address ----------------------------------------------------------------------------- SIP 5004 to 5082 Both(UDP&TCP) 192.168.1.100 RTP 5090 to 5100 UDP 192.168.1.100 That works OK. Now I want to divert that PSTN line from Asterisk to my Freeswitch server (192.168.1.101) So I changed in my router the ip addreese from 192.168.1.100 to 192.168.1.101 Application Start End Protocol IP Address ----------------------------------------------------------------------------- SIP 5004 to 5082 Both(UDP&TCP) 192.168.1.101 RTP 5090 to 5100 UDP 192.168.1.101 But.....when an external call comes in, it still goes to Asterisk. Am I on the wrong track or ....... (?) Rgds Henk