[asterisk-users] issue with NAT
rainer.piper at soho-piper.de
Mon Nov 3 06:47:37 CST 2014
Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens:
> First I am new to PBX so i might be doing something fundamentally
> 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
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.
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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