Have a quick look at this guide on NAT and SIP -
http://www.aocomputing.net/?p=3. This is the link given if you were to ask
this same question in the IRC channel...
--wcs
On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose <sixfourimp...@hotmail.com> wrote:
>
> Here is what i think the is helpful from wireshark
>
>
>
> OPTIONS sip:216.82.224.202 SIP/2.0
>
> Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK5aac5845;rport
>
> From: "Unknown" <sip:unkn...@mypublicip>;tag=as7b5287b3
>
> To: <sip:216.82.224.202>
>
> Contact: <sip:unkn...@mypublicip>
>
> Call-ID: 311d57516ef9649b7dfab93720393...@mypublicip
>
> CSeq: 102 OPTIONS
>
> User-Agent: Asterisk PBX
>
> Max-Forwards: 70
>
> Date: Wed, 21 Oct 2009 14:11:14 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>
> Supported: replaces
>
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK5aac5845;rport=5060
>
> From: "Unknown" <sip:unkn...@10.1.0.8 <sip%3aunkn...@10.1.0.8>
> >;tag=as7b5287b3
>
> To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.e340
>
> Call-ID: 311d57516ef9649b7dfab93720393...@10.1.0.8
>
> CSeq: 102 OPTIONS
>
> Server: Bandwidth.com TRM (bw7.gold.13)
>
> Content-Length: 0
>
>
>
> OPTIONS sip:216.82.224.202 SIP/2.0
>
> Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK2e9c787f;rport
>
> From: "Unknown" <sip:unkn...@mypublicip>;tag=as20c07cef
>
> To: <sip:216.82.224.202>
>
> Contact: <sip:unkn...@mypublicip>
>
> Call-ID: 09003fa1042464842df21c73339a1...@mypublicip
>
> CSeq: 102 OPTIONS
>
> User-Agent: Asterisk PBX
>
> Max-Forwards: 70
>
> Date: Wed, 21 Oct 2009 14:11:14 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>
> Supported: replaces
>
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK2e9c787f;rport=5060
>
> From: "Unknown" <sip:unkn...@10.1.0.8 <sip%3aunkn...@10.1.0.8>
> >;tag=as20c07cef
>
> To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.f75e
>
> Call-ID: 09003fa1042464842df21c73339a1...@10.1.0.8
>
> CSeq: 102 OPTIONS
>
> Server: Bandwidth.com TRM (bw7.gold.13)
>
> Content-Length: 0
>
>
>
> OPTIONS sip:216.82.224.202 SIP/2.0
>
> Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK4d179421;rport
>
> From: "Unknown" <sip:unkn...@mypublicip>;tag=as271c263c
>
> To: <sip:216.82.224.202>
>
> Contact: <sip:unkn...@mypublicip>
>
> Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@mypublicip
>
> CSeq: 102 OPTIONS
>
> User-Agent: Asterisk PBX
>
> Max-Forwards: 70
>
> Date: Wed, 21 Oct 2009 14:11:24 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>
> Supported: replaces
>
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK4d179421;rport=5060
>
> From: "Unknown" <sip:unkn...@10.1.0.8 <sip%3aunkn...@10.1.0.8>
> >;tag=as271c263c
>
> To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.25d4
>
> Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@10.1.0.8
>
> CSeq: 102 OPTIONS
>
> Server: Bandwidth.com TRM (bw7.gold.13)
>
> Content-Length: 0
>
>
>
> OPTIONS sip:216.82.224.202 SIP/2.0
>
> Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK61a8b371;rport
>
> From: "Unknown" <sip:unkn...@mypublicip>;tag=as3913f8ae
>
> To: <sip:216.82.224.202>
>
> Contact: <sip:unkn...@mypublicip>
>
> Call-ID: 05e50acb725d34f01bc78666422f5...@mypublicip
>
> CSeq: 102 OPTIONS
>
> User-Agent: Asterisk PBX
>
> Max-Forwards: 70
>
> Date: Wed, 21 Oct 2009 14:11:25 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>
> Supported: replaces
>
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK61a8b371;rport=5060
>
> From: "Unknown" <sip:unkn...@10.1.0.8 <sip%3aunkn...@10.1.0.8>
> >;tag=as3913f8ae
>
> To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.2790
>
> Call-ID: 05e50acb725d34f01bc78666422f5...@10.1.0.8
>
> CSeq: 102 OPTIONS
>
> Server: Bandwidth.com TRM (bw7.gold.13)
>
> Content-Length: 0
>
>
>
>
> ------------------------------
> From: sixfourimp...@hotmail.com
> To: asterisk-users@lists.digium.com
> Date: Wed, 21 Oct 2009 14:00:20 +0000
>
> Subject: Re: [asterisk-users] troubleshooting NAT
>
>
>
> > Date: Tue, 20 Oct 2009 21:02:29 -0500
> > From: asteriskl...@callthem.info
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] troubleshooting NAT
> >
> > if you're using SIP then you look at SIP headers ... SDP part
> > from INVITE's and 200 OK to INVITE. You check what IP/port is used for
> RTP
>
>
> Here is the SIP header that you see when you run the asterisk -r command.
>
> Reliably Transmitting (NAT) to 216.82.224.202:5060:
> OPTIONS sip:216.82.224.202 SIP/2.0
> Via: SIP/2.0/UDP ourpublicip:5060;branch=z9hG4bK42ef5901;rport
> From: "Unknown" <sip:unkn...@ourpublicip>;tag=as0186791c
> To: <sip:216.82.224.202>
> Contact: <sip:unkn...@ourpublicip>
> Call-ID: 52019c8970f8727a04fd79f0083cc...@ourpublicip
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 21 Oct 2009 13:33:36 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> Here is a debug from one of our phones calling an external number
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.46:5060
> ;branch=z9hG4bKb033ebc8ff0de35dc.650da238f5237c4a1;received=10.0.0.46
> From: "me" <sip:1...@10.1.0.8 <sip%3a...@10.1.0.8>>;tag=aa5daa3277
> To: "95457878" <sip:95457...@10.1.0.8 <sip%3a95457...@10.1.0.8>
> >;tag=as0b5e19fc
> Call-ID: 2edce254de2a77ab
> CSeq: 32330 CANCEL
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:95457...@10.1.0.8 <sip%3a95457...@10.1.0.8>>
> Content-Length: 0
>
>
> <------------>
> == Spawn extension (from-internal, 95457878, 4) exited non-zero on
> 'SIP/117-09c4fc20'
> -- Executing [...@from-internal:1] Macro("SIP/117-09c4fc20",
> "hangupcall") in new stack
> -- Executing [...@macro-hangupcall:1] GotoIf("SIP/117-09c4fc20",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [...@macro-hangupcall:4] GotoIf("SIP/117-09c4fc20",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [...@macro-hangupcall:7] GotoIf("SIP/117-09c4fc20",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [...@macro-hangupcall:9] Hangup("SIP/117-09c4fc20", "") in
> new stack
> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
> 'SIP/117-09c4fc20' in macro 'hangupcall'
> == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/117-09c4fc20'
>
> > and then you can try to get some packet dump with tcpdump/wireshark
>
> 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.
>
> 09:40:58.510750 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172
> 09:40:58.530758 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172
> 09:40:58.550762 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172
> 09:40:58.570770 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172
> 09:40:58.590775 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172
> 09:40:58.610781 IP 10.1.0.8.12036 > 10.1.0.46.hbci: UDP, length 172
> 09:40:58.625026 IP 10.1.0.46.sip > 10.1.0.8.sip: SIP, length: 348
> 09:40:58.625485 IP 10.1.0.8.sip > 10.1.0.46.sip: SIP, length: 417
> 09:40:58.625608 IP 10.1.0.8.sip > 10.1.0.46.sip: SIP, length: 435
> 09:40:58.679832 IP 10.1.0.46.sip > 10.1.0.8.sip: SIP, length: 334
>
>
>
>
>
> > and maybe configure your router
> > so it works.... it's the first thing to look for ...
>
> if the phone server can access the internet then shouldn't that mean the
> router has NAT setup correctly on it?
>
> >
> > you can also try to use the stun server ... asterisk has it built in
> > ...never used it but saw it's there
> >
> > Martin
> >
> > On Tue, Oct 20, 2009 at 1:32 PM, Ott Rose <sixfourimp...@hotmail.com>
> wrote:
> > > Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look
> at
> > > your install and they said we are having a NAT problem but didn'ttell
> me if
> > > it was with the asterisk conf or the Cisco ASA.
> > >
> > > ________________________________
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> >
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