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;tag=aa5daa3277 To: 95457878 sip:95457...@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 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. 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. http://clk.atdmt.com/GBL/go/171222985/direct/01/___ -- 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
Re: [asterisk-users] troubleshooting NAT
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;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;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;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;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 + 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;tag=aa5daa3277 To: 95457878 sip:95457...@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 Content-Length: 0 == Spawn extension (from-internal, 95457878, 4) exited non-zero on 'SIP/117-09c4fc20' -- Executing [...@from-internal:1] Macro(SIP/117-09c4fc20
Re: [asterisk-users] troubleshooting NAT
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 + 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
Re: [asterisk-users] troubleshooting NAT
i changed my sip_nat.conf file following the steps in that link. Still didn't work same debug info Date: Wed, 21 Oct 2009 10:33:18 -0500 From: wcse...@selbytech.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] troubleshooting NAT 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;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;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;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;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 + 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
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 and then you can try to get some packet dump with tcpdump/wireshark and maybe configure your router so it works it's the first thing to look for ... 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. 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
