[asterisk-users] issue with NAT

[Rainer Piper]

Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens:
> First I am new to PBX so i might be doing something fundamentally 
> wrong...
> That being said I got a FreePBX 32bit stable 6.12.65.
>
> I am having some issue with the NAT and sound, both phones are ringing 
> but there is sound, I had some talk on IRC:
>
> <[TK]D-Fender> Note for elfranne's situation, : 
> nat=force_rport,comedia" should have returned  the public IP the call 
> arrived on, but it is not.  Can anyone comment on why it wouldn't have 
> pulled it?
>
> A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu
>
>
>
>
Hi Tom,

you can add a STUN Server in your Linksys/SPA942-6.1.5(a) user agents.

read more about STUN at: http://www.voip-info.org/wiki/view/STUN
and there is a list of public STUN Server.

Regards

-- 
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:rainer at sip.soho-piper.de:5072 (pjsip-test)
XMPP: rainer at xmpp.soho-piper.de
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