[asterisk-users] issue with NAT

Rainer Piper 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 
> 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.


*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
Phone: +49 228 97167161
P2P: sip:rainer at sip.soho-piper.de:5072 (pjsip-test)
XMPP: rainer at xmpp.soho-piper.de
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141103/0f863e0a/attachment.html>

More information about the asterisk-users mailing list