I'm pretty sure I'm just missing something very simple here like a port forward. Let me know if you have any ideas! This setup completely works when both csipsiimple clients are inside my LAN and kamailio is configured with alias=192.168.1.LAN. The problem begins when I connect both clients from the WAN and configure kamailio with alias=66.41.221.WAN.
The problem is the callee continues to show the call as 'incoming call' after the call is answered and 2-way audio is established so eventually the callee times out.
Summary:: The callee and caller both establish 2-way audio and zrtp keys. TLS is working. Both SIP clients are using the OSTN wizard in csipsimple.
Please review my Debug logs: http://pastebin.com/9Rw7zSQ0
Hello,
maybe is better to just use:
listen=lanip advertise wanip
and let everyone connect via wanip.
Otheriwse, you have to play with set_advertised_address() and record_route_preset() depending on calls going from lan to lan, lan to wan, wan to lan and wan to wan.
You should send the ngrep of a broken call (from initial invite to the end), to see what is not working there.
Cheers, Daniel
On 11/3/13 6:08 PM, kamailio@AaronLux.com wrote:
I'm pretty sure I'm just missing something very simple here like a port forward. Let me know if you have any ideas! This setup completely works when both csipsiimple clients are inside my LAN and kamailio is configured with alias=192.168.1.LAN. The problem begins when I connect both clients from the WAN and configure kamailio with alias=66.41.221.WAN.
The problem is the callee continues to show the call as 'incoming call' after the call is answered and 2-way audio is established so eventually the callee times out.
Summary:: The callee and caller both establish 2-way audio and zrtp keys. TLS is working. Both SIP clients are using the OSTN wizard in csipsimple.
Please review my Debug logs: http://pastebin.com/9Rw7zSQ0
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