[Asterisk-Users] One sided call

[Woodoo People .pGa!]

Hi! I found, that there is 4 options for nat:
-no
-never
-yes
-always

no and never is ok
but sometimes yes, and sometimes always worked for me :-o
 
> I am having problem diagnosing a call problem. On both a Cisco phone and a
> Linksys 942 I am only getting one side of the call when connected over a WAN
> link or internet connection. I have set nat=yes and qualify in sip.conf and
> the phone registers fine. I can hear the other end, but they do not hear
> anything, no voice or dtmf. I found a tip about changing the RTP rate from
> .03 to .02 on Sipura phones to match Asterisk rate and did that. I also made
> sure the RTP range for the phone and the server was set to 10000 thru 20000.
> These phones work fine when on the same subnet as the server. The server
> shows the following message:


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