The asterisk is behind NAT.One client 222 is in the same private network. 222 client can call the remote user 101 who is bhind NAT too.But the 101 cant call the 222.When they change their account to login.Then 101 can call 222,but 222 cant call 101 too.Why ?I think it is the similar configure with 222 and 1o1.
Anybody can help me ? Thanks!
Please include some sip logs and config. do you have the setting “canreinvite=no” your sip device behind NAT?
What should i configure for canreinvite,yes or no?
What situation should be configured with yes,on the other hand,it should be configured with no?
At that time ,one is no and the other is yes.but it is the same result when changing account for testing. Is canreinvite relative to NAT?
in that case asterisk will be on the mids of the two channel. That is, the asterisk will copy RTP packet from caller and send it the callee, also copy the RTP from callee to and send it the caller. I suggest you check the SIP log of the device, if you found no problem then check the SDP. So, you can tell what’s going on.