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

[henkoegema]

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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> 

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