[asterisk-users] issue with NAT

[Matthew Jordan]

On Mon, Nov 3, 2014 at 6:58 AM, Rainer Piper <rainer.piper at soho-piper.de> wrote:
> Am 03.11.2014 um 13:47 schrieb 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
>
>
> the "add path header support in chan_sip" could help as well.
> look at   https://issues.asterisk.org/jira/browse/ASTERISK-16884
>
> [Test danes 202]
> ...
> ...
> nat=force_rport,comedia
> usepath=yes
> ...
> ...
>
> [test danes 203]
> ...
> ...
> nat=force_rport,comedia
> usepath=yes
> ...
> ...

Path support will only help if there are intermediary proxies, and
even then won't help with media (assuming OP meant 'no sound').

I could have missed it in the pastebin, but I didn't see a
request/response from Asterisk that was either sent to a private IP
address or contained a private IP address in the SDP. In the trace
that you provided, which request/response did you feel was in error?

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org