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i changed my sip_nat.conf file following the steps in that link. Still didn't work same debug info<br><br><hr id="stopSpelling">Date: Wed, 21 Oct 2009 10:33:18 -0500<br>From: wcselby@selbytech.com<br>To: asterisk-users@lists.digium.com<br>Subject: Re: [asterisk-users] troubleshooting NAT<br><br>Have a quick look at this guide on NAT and SIP - <a href="http://www.aocomputing.net/?p=3">http://www.aocomputing.net/?p=3</a>. This is the link given if you were to ask this same question in the IRC channel...<br><br>--wcs<br>
<br><br><div class="ecxgmail_quote">On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a>></span> wrote:<br><blockquote class="ecxgmail_quote" style="padding-left: 1ex;">
<div>
<br>Here is what i think the is helpful from wireshark <br><div class="ecxim"><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br></div>Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK5aac5845;rport<br><br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as7b5287b3<div class="ecxim">
<br><br>To: <sip:216.82.224.202><br><br></div>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 311d57516ef9649b7dfab937203932fb@mypublicip<div class="ecxim"><br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br>
<br>Max-Forwards: 70<br><br></div>Date: Wed, 21 Oct 2009 14:11:14 GMT<div class="ecxim"><br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br><br>Supported: replaces<br><br>Content-Length: 0<br><br><br>
<br></div><div class="ecxim">SIP/2.0 200 OK<br><br></div>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK5aac5845;rport=5060<br><br>From: "Unknown" <<a href="mailto:sip:Unknown@10.1.0.8">sip:Unknown@10.1.0.8</a>>;tag=as7b5287b3<br>
<br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.e340<br><br>Call-ID: <a href="mailto:311d57516ef9649b7dfab937203932fb@10.1.0.8">311d57516ef9649b7dfab937203932fb@10.1.0.8</a><br><br>
CSeq: 102 OPTIONS<br><br>Server: Bandwidth.com TRM (bw7.gold.13)<br><br>Content-Length: 0<div class="ecxim"><br><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br></div>Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK2e9c787f;rport<br>
<br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as20c07cef<div class="ecxim"><br><br>To: <sip:216.82.224.202><br><br></div>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 09003fa1042464842df21c73339a128a@mypublicip<div class="ecxim">
<br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br><br>Max-Forwards: 70<br><br></div>Date: Wed, 21 Oct 2009 14:11:14 GMT<div class="ecxim"><br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
<br>Supported: replaces<br><br>Content-Length: 0<br><br><br><br></div><div class="ecxim">SIP/2.0 200 OK<br><br></div>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK2e9c787f;rport=5060<br><br>From: "Unknown" <<a href="mailto:sip:Unknown@10.1.0.8">sip:Unknown@10.1.0.8</a>>;tag=as20c07cef<br>
<br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.f75e<br><br>Call-ID: <a href="mailto:09003fa1042464842df21c73339a128a@10.1.0.8">09003fa1042464842df21c73339a128a@10.1.0.8</a><br><br>
CSeq: 102 OPTIONS<br><br>Server: Bandwidth.com TRM (bw7.gold.13)<br><br>Content-Length: 0<div class="ecxim"><br><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br></div>Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK4d179421;rport<br>
<br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as271c263c<div class="ecxim"><br><br>To: <sip:216.82.224.202><br><br></div>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 30a25dbd0ed352e5159b0b6a767ea63b@mypublicip<div class="ecxim">
<br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br><br>Max-Forwards: 70<br><br></div>Date: Wed, 21 Oct 2009 14:11:24 GMT<div class="ecxim"><br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
<br>Supported: replaces<br><br>Content-Length: 0<br><br><br><br></div><div class="ecxim">SIP/2.0 200 OK<br><br></div>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK4d179421;rport=5060<br><br>From: "Unknown" <<a href="mailto:sip:Unknown@10.1.0.8">sip:Unknown@10.1.0.8</a>>;tag=as271c263c<br>
<br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.25d4<br><br>Call-ID: <a href="mailto:30a25dbd0ed352e5159b0b6a767ea63b@10.1.0.8">30a25dbd0ed352e5159b0b6a767ea63b@10.1.0.8</a><br><br>
CSeq: 102 OPTIONS<br><br>Server: Bandwidth.com TRM (bw7.gold.13)<br><br>Content-Length: 0<div class="ecxim"><br><br><br><br>OPTIONS sip:216.82.224.202 SIP/2.0<br><br></div>Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK61a8b371;rport<br>
<br>From: "Unknown" <sip:Unknown@mypublicip>;tag=as3913f8ae<div class="ecxim"><br><br>To: <sip:216.82.224.202><br><br></div>Contact: <sip:Unknown@mypublicip><br><br>Call-ID: 05e50acb725d34f01bc78666422f5acd@mypublicip<div class="ecxim">
<br><br>CSeq: 102 OPTIONS<br><br>User-Agent: Asterisk PBX<br><br>Max-Forwards: 70<br><br></div>Date: Wed, 21 Oct 2009 14:11:25 GMT<div class="ecxim"><br><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
<br>Supported: replaces<br><br>Content-Length: 0<br><br><br><br></div><div class="ecxim">SIP/2.0 200 OK<br><br></div>Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK61a8b371;rport=5060<br><br>From: "Unknown" <<a href="mailto:sip:Unknown@10.1.0.8">sip:Unknown@10.1.0.8</a>>;tag=as3913f8ae<br>
<br>To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.2790<br><br>Call-ID: <a href="mailto:05e50acb725d34f01bc78666422f5acd@10.1.0.8">05e50acb725d34f01bc78666422f5acd@10.1.0.8</a><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>From: <a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a><div class="ecxim">
<br>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br></div>Date: Wed, 21 Oct 2009 14:00:20 +0000<div><div></div><div class="h5"><br>Subject: Re: [asterisk-users] troubleshooting NAT<br>
<br>
<br><br>> Date: Tue, 20 Oct 2009 21:02:29 -0500<br>> From: <a href="mailto:asterisklist@callthem.info">asterisklist@callthem.info</a><br>> To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><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 <a href="http://216.82.224.202:5060">216.82.224.202:5060</a>:<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" <<a href="mailto:sip:117@10.1.0.8">sip:117@10.1.0.8</a>>;tag=aa5daa3277<br>
To: "95457878" <<a href="mailto:sip:95457878@10.1.0.8">sip:95457878@10.1.0.8</a>>;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: <<a href="mailto:sip:95457878@10.1.0.8">sip:95457878@10.1.0.8</a>><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 <<a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a>> 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>
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