[asterisk-users] Still sipping frustration - only getting state ACK

[Julien Claassen]

Hello everyone!
   I still am not much further along with my sip calling. I changed my sip.conf 
taking suggestions from the net (voip-info.org in particular). I changed 
iptel's position from friend to peer. I turned on and off nat, I chose 
different codecs in first place, entered my outward IP as fromdomain and 
uncommented the register directive with correct values.
   All I get is two registrations now, but no calls.  get a registration effort 
every 225secs and it succeeds. But when I make a call;
channel originate sip/iptel-out/echo at iptel.org Application playback 
vm/net_ring
   The call is onlyleft in state ACK for a while. Then asterisk tells me, that 
it is destroying the sip dialog (long ID) INVITE.
   Question: Might it be a problem, that my system only knows itself as 
192.168.*. Do I need to set something else than externip?
   Might it be, that my router really blocks certain ports? I can't check it, 
since it's heavily javascript based and, since I'm blind and the accessibility 
software for the GUI never really worked on this distro, I don't have a 
browser to look at it.
   Do I need to forward port 5060 to my machine specifically (like it is needed 
for SSH's port 22), or is the mechanism based on: I talk first and the sever 
gets back to me based on that.
   This configuration worked for googletalk. I admit, there were problems, but 
calls were coming through from both sides.
   Please can someone help me clear up this mess. I'm completely frustrated and 
don't know what else to do, where else to look.
   Kindly yours and thanks in advance
             JUlien

--------
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