[Asterisk-Users] help with SIP softphones

[Daniel Flickinger]

Hello,

   I'm new to Asterisk, and am trying to get the basic features under my belt 
until I move on to the more advanced ones.  Currently I have two softphones 
registered with my * server on my network, and one of the phones can call the 
other just fine, but when I try to call from the other phone, I get a message 
on the phone saying that the call was terminated for reasons unknown.  When I 
register the phones on the asterisk server, the phone that can dial to other 
phones is always assigned to port 5061, whereas the other ones are on port 
5060.  My guess is that my problem has something to do with the dialing phone 
being on port 5061.  When I try to dial manually from another phone to the 
phone on port 5061 with the port specified, I still can't reach it.  Any help 
on why this is occurring will be very much appreciated.  Thanks

daniel