[Asterisk-Users] Grandstream and Snome remote sip stops taking calls

[Jason]

I have remote users that are setup to sip into the Asterisk server.
Problem is that if you call there extension after they have been registered
For a while there phones don't ring.
If I do a sip show peers they can be seen as registered in.
Also the user can dial out.
If they reset the phone they can receive calls.
This seems to be more of an issue with the Grand stream phones.

The Grandstream has these two settings I am un sure of.
NAT Traversal (STUN):  currently set to no
SUBSCRIBE for MWI: currently set to no

Any ideas?

-Jason