Hey Everyone, I'm running into a weird issue, and hopefully someone can assist me in finding out what's going on.
I'm running Astlinux 0.7 on a box serving as my router, asterisk box and openvpn server (and a few other things) and I've run into a seemingly very unusual issue. I have an ATA setup behind my Astlinux box that is remotely connecting to a fax server at work running freeswitch. It will be using t.38 and connecting directly to this server, completely bypassing my Astlinux box (outside of it serving as a router+nat). It registers fine, and can make and receive calls. The issue occurs when the rtp is being setup. I discovered the problem because I couldn't get faxes to work at all, even though everyone else had no problem. Even other people behind a similar setup (though not running this version of Astlinux). I noticed in my nat table that my rtp was going from the ATA to 19.226.0.0. Very very unusual. So I started running tcpdump on both eth0 and eth1 (wan and lan respectively). It seems like something in my astlinux box is modifying the contents of the sip packets and changing the rtp IP addresses. Everything from the server on eth0 looks perfect, as does everything from the ATA on eth1. But when I look at eth1 from the server, the rtp address is being set to 19.226.0.0. And when I look at eth0 from the ATA, my rtp address is being set to 10.200.143.207. The specific SIP packet capture in question here from the egress interfaces: (with some slight IP/hostname/phone number obfuscations to protect the innocent ;) ) --==-- 15:23:38.985020 IP (tos 0x68, ttl 249, id 9380, offset 0, flags [none], proto: UDP (17), length: 726) my.public.address.5090 > remote.server.address.5060: SIP, length: 698 SIP/2.0 200 OK To: James Babiak FAX <sip:fax_xx...@remote.server.address>;tag=63e72d1726ec8327o0 From: <sip:5551...@remote.server.address>;tag=0aDa2FSmFDScc Call-ID: f12527d7-85e8c...@172.20.0.145 CSeq: 126063973 INVITE Via: SIP/2.0/UDP remote.server.address;branch=z9hG4bKKeXpa1my0UF0r Contact: James Babiak FAX <sip:fax_xx...@remote.server.address:5090> Server: Linksys/SPA3102-5.1.10(GW) Content-Length: 269 Content-Type: application/sdp v=0 o=- 128599 128599 IN IP4 10.200.143.207 s=- c=IN IP4 10.200.143.207 t=0 0 m=image 16434 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy --==-- and --==-- 15:19:24.575373 IP (tos 0x20, ttl 50, id 57433, offset 0, flags [none], proto: UDP (17), length: 1084) remote.server.address.5060 > SipuraSPA.routed.com.5090: SIP, length: 1056 INVITE sip:fax_xx...@172.20.0.145:5090 SIP/2.0 Via: SIP/2.0/UDP 38.101.17.105;rport;branch=z9hG4bKFarK3mHgcrpNa Max-Forwards: 70 From: <sip:xx...@remote.server.address>;tag=e416y1XHNr42g To: James Babiak FAX <sip:fax_xx...@remote.server.address>;tag=74f688c7b624f7dao0 Call-ID: 91cd090b-14d4d...@172.20.0.145 CSeq: 126063846 INVITE Contact: <sip:5551...@remote.server.address:5060;transport=udp> User-Agent: Star2Star Media Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 316 X-FS-Support: update_display v=0 o=Sonus_UAC 2924581275921585578 5556846930163024566 IN IP4 19.226.0.0 s=SIP Media Capabilities c=IN IP4 19.226.0.0 t=0 0 m=image 11692 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy --==-- But like I said, the corresponding packets on the ingress interface does not reflect those above IP addresses. They have the proper ones. So something internal is changing them. I've spent hours trying to find some firewall rule/setting I needed to change, and even went so far as to disable everything that wasn't 'standard', but nothing works. I've tried changing SIP ports for the ATA, setting up port forwarding, etc. etc., but still no go. The remote freeswitch side seems to be ignoring my invalid IP, since the ATA still receives inbound audio, just no outbound audio. Everything else, IP's included, in the SIP packets are fine. It only breaks after the call is setup. Any ideas? Thanks -James ------------------------------------------------------------------------------ Throughout its 18-year history, RSA Conference consistently attracts the world's best and brightest in the field, creating opportunities for Conference attendees to learn about information security's most important issues through interactions with peers, luminaries and emerging and established companies. http://p.sf.net/sfu/rsaconf-dev2dev _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.