Re: [asterisk-users] issue with NAT
On Mon, Nov 3, 2014 at 6:58 AM, Rainer Piper 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with NAT
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:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with NAT
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 -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
