4 Nov
2013
4 Nov
'13
9:47 a.m.
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(a)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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