Scenario: a SIP server running Elastix (Asterisk and FreePBX) with SIP-clients behind NAT. Outgoing calls from the clients (which where CounterPath Xten and Siemens Gigaset C530ip, i.e. completely different brands) works well, but incoming calls are disconnected after 8-10 seconds.
Googleing the problem suggests the problem is caused by a SIP REINVITE on the client side, which can’t happen because of NAT. During testing, I discovered that calls between SIP-phones connected to the same SIP PBX worked fine, even between different locations. Only calls that originated externally would be interrupted.
This made me focus on the trunks instead and I found out that the problem indeed was a REINVITE but it occured on the trunk side.
The solution was to add the following lines to the PEER DETAILS and USER DETAILS on the trunk configuration in the SIP server: