[Freeswitch-users] How to divert a virtual PSTN line to another server ?

[Henk Oegema]

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