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
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