<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Verdana
}
--></style>
</head>
<body class='hmmessage'>
<br>Here is what i think the is helpful from wireshark <br><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br>Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK5aac5845;rport<br><br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as7b5287b3<br><br>To: <sip:216.82.224.202><br><br>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 311d57516ef9649b7dfab937203932fb@mypublicip<br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br><br>Max-Forwards: 70<br><br>Date: Wed, 21 Oct 2009 14:11:14 GMT<br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br><br>Supported: replaces<br><br>Content-Length: 0<br><br><br><br>SIP/2.0 200 OK<br><br>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK5aac5845;rport=5060<br><br>From: "Unknown" <sip:Unknown@10.1.0.8>;tag=as7b5287b3<br><br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.e340<br><br>Call-ID: 311d57516ef9649b7dfab937203932fb@10.1.0.8<br><br>CSeq: 102 OPTIONS<br><br>Server: Bandwidth.com TRM (bw7.gold.13)<br><br>Content-Length: 0<br><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br>Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK2e9c787f;rport<br><br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as20c07cef<br><br>To: <sip:216.82.224.202><br><br>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 09003fa1042464842df21c73339a128a@mypublicip<br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br><br>Max-Forwards: 70<br><br>Date: Wed, 21 Oct 2009 14:11:14 GMT<br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br><br>Supported: replaces<br><br>Content-Length: 0<br><br><br><br>SIP/2.0 200 OK<br><br>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK2e9c787f;rport=5060<br><br>From: "Unknown" <sip:Unknown@10.1.0.8>;tag=as20c07cef<br><br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.f75e<br><br>Call-ID: 09003fa1042464842df21c73339a128a@10.1.0.8<br><br>CSeq: 102 OPTIONS<br><br>Server: Bandwidth.com TRM (bw7.gold.13)<br><br>Content-Length: 0<br><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br>Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK4d179421;rport<br><br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as271c263c<br><br>To: <sip:216.82.224.202><br><br>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 30a25dbd0ed352e5159b0b6a767ea63b@mypublicip<br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br><br>Max-Forwards: 70<br><br>Date: Wed, 21 Oct 2009 14:11:24 GMT<br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br><br>Supported: replaces<br><br>Content-Length: 0<br><br><br><br>SIP/2.0 200 OK<br><br>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK4d179421;rport=5060<br><br>From: "Unknown" <sip:Unknown@10.1.0.8>;tag=as271c263c<br><br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.25d4<br><br>Call-ID: 30a25dbd0ed352e5159b0b6a767ea63b@10.1.0.8<br><br>CSeq: 102 OPTIONS<br><br>Server: Bandwidth.com TRM (bw7.gold.13)<br><br>Content-Length: 0<br><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br>Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK61a8b371;rport<br><br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as3913f8ae<br><br>To: <sip:216.82.224.202><br><br>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 05e50acb725d34f01bc78666422f5acd@mypublicip<br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br><br>Max-Forwards: 70<br><br>Date: Wed, 21 Oct 2009 14:11:25 GMT<br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br><br>Supported: replaces<br><br>Content-Length: 0<br><br><br><br>SIP/2.0 200 OK<br><br>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK61a8b371;rport=5060<br><br>From: "Unknown" <sip:Unknown@10.1.0.8>;tag=as3913f8ae<br><br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.2790<br><br>Call-ID: 05e50acb725d34f01bc78666422f5acd@10.1.0.8<br><br>CSeq: 102 OPTIONS<br><br>Server: Bandwidth.com TRM (bw7.gold.13)<br><br>Content-Length: 0<br><br><br><br><br><hr id="stopSpelling">From: sixfourimpala@hotmail.com<br>To: asterisk-users@lists.digium.com<br>Date: Wed, 21 Oct 2009 14:00:20 +0000<br>Subject: Re: [asterisk-users] troubleshooting NAT<br><br>
<style>
.ExternalClass .ecxhmmessage P
{padding:0px;}
.ExternalClass body.ecxhmmessage
{font-size:10pt;font-family:Verdana;}
</style>
<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. Get Windows Live Hotmail Free. <a href="http://clk.atdmt.com/GBL/go/171222985/direct/01/">Sign up now.</a> <br /><hr />Hotmail: Free, trusted and rich email service. <a href='http://clk.atdmt.com/GBL/go/171222984/direct/01/' target='_new'>Get it now.</a></body>
</html>