[asterisk-users] issue with NAT

[Rainer Piper]

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
...
...


> -- 
> *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
>
>


-- 
*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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