[Asterisk-Users] SIP client<->NAT<->Asterisk<->NAT<->SIP client. only works with canreinvite=no.

[WipeOut .]

> Can anybody explain me what does canreinvite=yes really does?
> 
Not sure how technical an answer you want becasue it look slike you know whats going on but as I unterstand it "canreinvite=no" tells the UA that reinvite is not supported and so causes all the RTP traffic to be routed via the * server.. I played with many nat settings and port forwarding settings and it ended up that "canreinvite=no" was the solution to my problems as well.. the downside is that it requires more bandwidth at the central site but the plus side is that it works through NAT..


> Any ideas on the client A to C (same LAN, same NAT box, unique outside 
> IP, same * server)?
> 
Only thing that springs to mind is to install another * box internally and then use IAX to connect the internal * box to the external one.. then the internal phone will call each other without crossing the NAT..

Later..
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