[Asterisk-Users] sip to sip

[Quintin]

Hi 

 

I'm trying to put up an sip pbx system for my company but i'm getting some
problems when I'm trying to call from server ( branch A ) to server ( branch
B ).

 

This is my extentions.conf :

 

exten => 3003,1,Dial,SIP/3003 at 192.168.0.200

 

________________________________________________________

 

 

And this is what I get when I try to dial that user in branch B

 

_________________________________________________________

 

    -- Executing Dial("SIP/5001-66b1", "SIP/3003 at 192.168.0.200") in new
stack

    -- Called 3003 at 192.168.0.200

    -- Got SIP response 404 "Not Found" back from 192.168.0.200

    -- SIP/192.168.0.200-e638 is circuit-busy

  == Everyone is busy/congested at this time (1:0/1/0)

  == Auto fallthrough, channel 'SIP/5001-66b1' status is 'CONGESTION'

 

Both servers are exactly the same... 

 

What can the problem be, that branch B server doesn't route the call through

 

Thx

Quintin

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