[Asterisk-Users] Asterisk behind LinkSys NAT Routing

[Philipp von Klitzing]

Hi!

> I don't think that is what keeping the original poster's system from
> working. The issue is "one" extension is configured for canreinvite=no
> and the other is canreinvite=yes. One extension believes all RTP must
> be passed through * while the other is attempting to negotiate a
> phone-to-phone RTP session, thus dropping the audio. 

Are you sure this is 100% correct? I have some doubts since:

- you'd have to consider all possible connection permutations between all 
clients and then set canreinvite= accordingly, which doesn't sound like 
it makes much sense

- sip.conf is for * only, the data are not seen or read by the SIP UA 
themselves. Thus it would appear that it is up to * to permit/not permit 
a reinvite between the two UAs

So bascially from my understanding things work like this: Once one of the 
SIP call parties has a canreinvite=no it won't matter what the other 
party's setting looks like, RTP traffic will travel through * anyway.

Am I wrong here?
Philipp