I realise this is a development forum, not support, but people who use this forum understand the details in SIP, VoIP & NAT traversal - and I hope will know the answer.
I have an E63 with up to date firmware, linked to an Asterisk 1.6 PBX via an Ad-hoc WiFi network. There is NAT involved. The E63 registers without problems & stays registered.
If you place a call OUT from the E63, there are no issues.
If you place a call IN to the E63, it rings & you can answer the call. You cannot hear anything from the person calling IN, but they can hear everything from the E63.
I'm assuming this is a NAT traversal RTP issue, but I'm struggling to find how to fix it.
I've tried UDP & TCP. I've tried reducing TCP & UDP refresh rates down to 4.
I've tried with & without specifying the Asterisk server as a Proxy server.
I 'think' I need to use a STUN server to fix this - but I've tried searching for Stun server and Asterisk 1.6 and it seems that there is no STUN implementation in Asterisk 1.6
This works on a Windows mobile phone using a SIP client so I'm hoping this is just a config issue for the Nokia client. I'd welcome any suggestions.