[SR-Users] Kamailio callee timeout problem

Daniel-Constantin Mierla miconda at gmail.com
Mon Nov 4 09:47:59 CET 2013


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 at 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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-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Trainings - Berlin, Nov 25-28
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