From my recent adventures with SIP, the SIP server your are connecting to, may have changed a bit.
There is 2 types of connection when doing a SIP call : SIP server connection and RTP server connection
Be it for SIP (registration, phone numbers presentation, call signaling, codec compatibility exchanges before RTP voice streams startup, and RTP servers and ports that should be used for streaming your voice) or for RTP streams themselves (voice), some server will just answer to the source of the UDP datagrams sent by your SPA112 (most easy ones). You connect to the server, so the server just has to answer when he has something to say. When you have an ingoing call, the server is inviting you through the SIP connection, giving the RTP server address and port to connect for taking the voice (and the RTP server is even waiting for you to connect).
Others ones will want your SPA112 to explicitely give your external IP/Port while connecting to it, for SIP and/or for RTP streams.
In this purpose, you can try to use Voice -> SIP -> NAT Support Parameters :
- Handle VIA received: YES
- Handle VIA rport:YES
This way, after the first packet is exchanged, the SPA112 will know it's address:port thanks to the "VIA" header of the servers's messages. STUN servers are an alternative to do the same using another server.
In order to use it, you may need to go to Voice -> Line1 -> NAT Settings -> NAT Mapping Enable: YES.
After rebooting to be sure the changes are applied on a fresh new connection to the server, if it doesn't work, maye be the voice/RTP server is trying to reach you first (then fail and stops talking to you) just before you begin to talk with it (but to late, the ingoing RTP stream is already closed) Just to figure out what is the problem, you can try to make your RTP port range permanently reachable through your NAT, so that this cannot happen.
This port range is defined into Voice -> SIP -> RTP Parameters -> RTP Port Min / RTP Port Max.
If it's opened, the RTP stream from the server will succeed to reach your SPA112 even before the phone begins to ring.
Keep us informed !
PS : if you are loosing some incoming calls (sometimes not ringing), try to enable Keep Alives : sometimes it's done at server side (OPTIONS messages repeatedly sent by the server once you are connected to it) so that your NAT knows the connection is still active. If the new server doesn't do so, you will have to ask the SPA112 to do so. But this is not the problem you are describing here, so that's just in case if it happens.
