[asterisk-users] troubleshooting NAT

Wed Oct 21 09:59:35 CDT 2009

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:Unknown at mypublicip>;tag=as7b5287b3

To: <sip:216.82.224.202>

Contact: <sip:Unknown at mypublicip>

Call-ID: 311d57516ef9649b7dfab937203932fb at 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:Unknown at 10.1.0.8>;tag=as7b5287b3

To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.e340

Call-ID: 311d57516ef9649b7dfab937203932fb at 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:Unknown at mypublicip>;tag=as20c07cef

To: <sip:216.82.224.202>

Contact: <sip:Unknown at mypublicip>

Call-ID: 09003fa1042464842df21c73339a128a at 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:Unknown at 10.1.0.8>;tag=as20c07cef

To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.f75e

Call-ID: 09003fa1042464842df21c73339a128a at 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:Unknown at mypublicip>;tag=as271c263c

To: <sip:216.82.224.202>

Contact: <sip:Unknown at mypublicip>

Call-ID: 30a25dbd0ed352e5159b0b6a767ea63b at 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:Unknown at 10.1.0.8>;tag=as271c263c

To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.25d4

Call-ID: 30a25dbd0ed352e5159b0b6a767ea63b at 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:Unknown at mypublicip>;tag=as3913f8ae

To: <sip:216.82.224.202>

Contact: <sip:Unknown at mypublicip>

Call-ID: 05e50acb725d34f01bc78666422f5acd at 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:Unknown at 10.1.0.8>;tag=as3913f8ae

To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.2790

Call-ID: 05e50acb725d34f01bc78666422f5acd at 10.1.0.8

CSeq: 102 OPTIONS

Server: Bandwidth.com TRM (bw7.gold.13)

Content-Length: 0

From: sixfourimpala at hotmail.com
To: asterisk-users at 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: asterisklist at callthem.info
> To: asterisk-users at 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:Unknown at ourpublicip>;tag=as0186791c
To: <sip:216.82.224.202>
Contact: <sip:Unknown at ourpublicip>
Call-ID: 52019c8970f8727a04fd79f0083cce21 at 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:117 at 10.1.0.8>;tag=aa5daa3277
To: "95457878" <sip:95457878 at 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:95457878 at 10.1.0.8>
Content-Length: 0

<------------>
  == Spawn extension (from-internal, 95457878, 4) exited non-zero on 'SIP/117-09c4fc20'
    -- Executing [h at from-internal:1] Macro("SIP/117-09c4fc20", "hangupcall") in new stack
    -- Executing [s at macro-hangupcall:1] GotoIf("SIP/117-09c4fc20", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s at macro-hangupcall:4] GotoIf("SIP/117-09c4fc20", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s at macro-hangupcall:7] GotoIf("SIP/117-09c4fc20", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s at 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 <sixfourimpala at 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.
> >
> > ________________________________
> > Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up
> > now.
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
http://clk.atdmt.com/GBL/go/171222984/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/3229f7ad/attachment.htm 