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<br><br>> Date: Tue, 20 Oct 2009 21:02:29 -0500<br>> From: asterisklist@callthem.info<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] troubleshooting NAT<br>> <br>> if you're using SIP then you look at SIP headers ... SDP part<br>> from INVITE's and 200 OK to INVITE. You check what IP/port is used for RTP<br><br><br>Here is the SIP header that you see when you run the asterisk -r command.<br><br>Reliably Transmitting (NAT) to 216.82.224.202:5060:<br>OPTIONS sip:216.82.224.202 SIP/2.0<br>Via: SIP/2.0/UDP ourpublicip:5060;branch=z9hG4bK42ef5901;rport<br>From: "Unknown" <sip:Unknown@ourpublicip>;tag=as0186791c<br>To: <sip:216.82.224.202><br>Contact: <sip:Unknown@ourpublicip><br>Call-ID: 52019c8970f8727a04fd79f0083cce21@ourpublicip<br>CSeq: 102 OPTIONS<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Wed, 21 Oct 2009 13:33:36 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Length: 0<br><br><br>Here is a debug from one of our phones calling an external number<br><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.1.0.46:5060;branch=z9hG4bKb033ebc8ff0de35dc.650da238f5237c4a1;received=10.0.0.46<br>From: "me" <sip:117@10.1.0.8>;tag=aa5daa3277<br>To: "95457878" <sip:95457878@10.1.0.8>;tag=as0b5e19fc<br>Call-ID: 2edce254de2a77ab<br>CSeq: 32330 CANCEL<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:95457878@10.1.0.8><br>Content-Length: 0<br><br><br><------------><br> == Spawn extension (from-internal, 95457878, 4) exited non-zero on 'SIP/117-09c4fc20'<br> -- Executing [h@from-internal:1] Macro("SIP/117-09c4fc20", "hangupcall") in new stack<br> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/117-09c4fc20", "1?skiprg") in new stack<br> -- Goto (macro-hangupcall,s,4)<br> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/117-09c4fc20", "1?skipblkvm") in new stack<br> -- Goto (macro-hangupcall,s,7)<br> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/117-09c4fc20", "1?theend") in new stack<br> -- Goto (macro-hangupcall,s,9)<br> -- Executing [s@macro-hangupcall:9] Hangup("SIP/117-09c4fc20", "") in new stack<br> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/117-09c4fc20' in macro 'hangupcall'<br> == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/117-09c4fc20'<br><br>> and then you can try to get some packet dump with tcpdump/wireshark<br><br>if am ssh into the server and run tcpdump not port 22. i get normal LAN traffic until i make a call. then i get a ton of this. .8 is the phoneserver and .46 is one of the phones. i haven't done wireshark because I haven't looked up how to take the tcpdump and import it into wireshark. <br><br>09:40:58.510750 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172<br>09:40:58.530758 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172<br>09:40:58.550762 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172<br>09:40:58.570770 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172<br>09:40:58.590775 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172<br>09:40:58.610781 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172<br>09:40:58.625026 IP 10.1.0.46.sip > 10.1.0.8.sip: SIP, length: 348<br>09:40:58.625485 IP 10.1.0.8.sip > 10.1.0.46.sip: SIP, length: 417<br>09:40:58.625608 IP 10.1.0.8.sip > 10.1.0.46.sip: SIP, length: 435<br>09:40:58.679832 IP 10.1.0.46.sip > 10.1.0.8.sip: SIP, length: 334<br><br><br><br><br><br>> and maybe configure your router<br>> so it works.... it's the first thing to look for ...<br><br>if the phone server can access the internet then shouldn't that mean the router has NAT setup correctly on it? <br><br>> <br>> you can also try to use the stun server ... asterisk has it built in<br>> ...never used it but saw it's there<br>> <br>> Martin<br>> <br>> On Tue, Oct 20, 2009 at 1:32 PM, Ott Rose <sixfourimpala@hotmail.com> wrote:<br>> > Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at<br>> > your install and they said we are having a NAT problem but didn'ttell me if<br>> > it was with the asterisk conf or the Cisco ASA.<br>> ><br>> > ________________________________<br>> > Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up<br>> > now.<br>> > _______________________________________________<br>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> ><br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> ><br>> <br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> <br /><hr />Your E-mail and More On-the-Go. 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