I'm still looking for a solution. :working:
henkoegema wrote: > > 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 > > > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---tp19335394p19366567.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
