[asterisk-users] issue with NAT
rainer.piper at soho-piper.de
Mon Nov 3 06:58:16 CST 2014
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
>> 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.
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]
[test danes 203]
> *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
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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