[Asterisk-Users] RE:read .what else to do ?

[Taiwo Oluyemi]

 Thanks .Find My replies in between your lines   
"Please note that recent IOS has SIP NAT traversal turned on by default.
    
I believe that it only supports internal UA / external server.
    
Since you also want the opposite, you should probably turn it off:
    
no ip nat service sip tcp port 5060
    
no ip nat service sip udp port 5060
    
Some IOS versions will even crash on SIP behind NAT.  See
    
http://lists.digium.com/pipermail/asterisk-users/2004-January/033718.html”
    
Hope your comment above will not affect the Natting configuration on my router . What will be the effect of me turning it off ?
    
 
    
 
    
“You don't say what's failing.  "Make calls outside our LAN" sounds like
    
you are trying to call using a VoIP provider that Asterisk registers
    
with.  But "your remote SIP phones" is something different; which of
    
the above are failing?  Are the registrations successful?  Is it just
    
the RTP that's not working (in which case the called phone will still
    
ring)?  If not, what error or timeout is reported?”
    
We are not using a VOIP service provider, we use asterisks server behind a nat device (Cisco router), and the asterisk server is connected to an E1 link. We can make and receive calls on pc with xten  that is on the same private LAN with the asterisk server
    
 
    
“If * verbose and/or debug logs don't show precisely what is going 
    
wrong,
    
use Ethereal (on both sides of the router if necessary) to see what
    
is happening.”
    
How can I use verbose and/or debug logs and Ethereal
  

		
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