VoIP Mailing List Archives :: View topic - [asterisk-users] issue with NAT

Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community

[asterisk-users] issue with NAT

VoIP Mailing List Archives Forum Index -> Asterisk Users

View previous topic :: View next topic

Author

Message

tom at plustel.dk
Guest

Posted: Mon Nov 03, 2014 7:28 am Post subject: [asterisk-users] issue with NAT

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 -- _____________________________________________________________________ -- 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

rainer.piper at soho-p...
Guest

Posted: Mon Nov 03, 2014 7:48 am Post subject: [asterisk-users] issue with NAT

Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens:

Quote:

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: [url=sip:rainer@sip.soho-piper.de:5072]sip:rainer@sip.soho-piper.de:5072[/url] (pjsip-test)
XMPP: rainer@xmpp.soho-piper.de (rainer@xmpp.soho-piper.de)

rainer.piper at soho-p...
Guest

Posted: Mon Nov 03, 2014 7:58 am Post subject: [asterisk-users] issue with NAT

Am 03.11.2014 um 13:47 schrieb Rainer Piper:

Quote:

Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens:

Quote:

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
...
...

Quote:

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: [url=sip:rainer@sip.soho-piper.de:5072]sip:rainer@sip.soho-piper.de:5072[/url] (pjsip-test)
XMPP: rainer@xmpp.soho-piper.de (rainer@xmpp.soho-piper.de)

mjordan at digium.com
Guest

Posted: Mon Nov 03, 2014 9:15 am Post subject: [asterisk-users] issue with NAT

On Mon, Nov 3, 2014 at 6:58 AM, Rainer Piper <rainer.piper@soho-piper.de> wrote:

Quote:

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

Display posts from previous:

	VoIP Mailing List Archives Forum Index -> Asterisk Users	All times are GMT - 5 Hours
Page 1 of 1

You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum