I'm sure that Raul had a good point that I'm crazy for doing this, but right now unfortunately I do not have a lot of options, and believe with some solid logic in the config can resolve this issue. Right now I appear to be having an ACK come back from a 200 that turns into an endless loop. I'm using
advertised_address="75.101.136.125" U 10.250.7.164:5060 -> 76.102.118.209:3724 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.102:3724 ;received=76.102.118.209;branch=z9hG4bK-d8754z-ff34ad60b13c376a-1---d8754z-;rport=3724. Call-ID: NTU3NTE3NmZiNjVkNGYzNWEzYWVhZWQ4MjhhYjczN2E.. CSeq: 2 INVITE. Contact: <sip:[email protected]:11386>. From: "Daniel Goepp" <sip:[email protected] <sip%[email protected]> >;tag=ac7fa632. To: "2001" <sip:[email protected] <sip%[email protected]> >;tag=8f7b7e78f8cb0eca. Record-Route: <sip:75.101.136.125;lr=on>. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY. Server: TANDBERG/257 (TE2.0.0.191892). Supported: replaces,100rel,timer,gruu,path,outbound,com.tandberg.sdp.extensions.v1. Session-Expires: 500; refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 217. . v=0. o=tandberg 103 1 IN IP4 192.168.1.101. s=-. c=IN IP4 192.168.1.101. b=CT:64. t=0 0. m=audio 2330 RTP/AVP 0 101. b=TIAS:64000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 76.102.118.209:3724 -> 10.250.7.164:5060 ACK sip:[email protected]:11386 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102:3724 ;branch=z9hG4bK-d8754z-6d520059c54f5b74-1---d8754z-;rport. Max-Forwards: 70. Route: <sip:75.101.136.125;lr>. Contact: <sip:[email protected]:3724>. To: "2001"<sip:[email protected] <sip%[email protected]> >;tag=8f7b7e78f8cb0eca. From: "Daniel Goepp"<sip:[email protected] <sip%[email protected]> >;tag=ac7fa632. Call-ID: NTU3NTE3NmZiNjVkNGYzNWEzYWVhZWQ4MjhhYjczN2E.. CSeq: 2 ACK. Proxy-Authorization: Digest username="1001",realm="vidtel.com ",nonce="4ad65ba02105a03bfdc0e3839160a42b5e1d90ac",uri="sip:[email protected]<sip%[email protected]> ",response="0eea648b121320214ab8ec908eb97446",algorithm=MD5. User-Agent: eyeBeam release 1104g stamp 54685. Content-Length: 0. . U 10.250.7.164:5060 -> 75.101.136.125:5060 ACK sip:[email protected]:11386 SIP/2.0. Via: SIP/2.0/UDP 75.101.136.125;branch=z9hG4bK3128.7db9df05.2. Via: SIP/2.0/UDP 192.168.1.102:3724 ;received=76.102.118.209;branch=z9hG4bK-d8754z-6d520059c54f5b74-1---d8754z-;rport=3724. Max-Forwards: 69. Route: <sip:75.101.136.125;lr>. Contact: <sip:[email protected]:3724>. To: "2001"<sip:[email protected] <sip%[email protected]> >;tag=8f7b7e78f8cb0eca. From: "Daniel Goepp"<sip:[email protected] <sip%[email protected]> >;tag=ac7fa632. Call-ID: NTU3NTE3NmZiNjVkNGYzNWEzYWVhZWQ4MjhhYjczN2E.. CSeq: 2 ACK. Proxy-Authorization: Digest username="1001",realm="vidtel.com ",nonce="4ad65ba02105a03bfdc0e3839160a42b5e1d90ac",uri="sip:[email protected]<sip%[email protected]> ",response="0eea648b121320214ab8ec908eb97446",algorithm=MD5. User-Agent: eyeBeam release 1104g stamp 54685. Content-Length: 0. . -dg On Wed, Oct 7, 2009 at 11:20 PM, Bogdan-Andrei Iancu <[email protected] > wrote: > Hi Daniel, > > Try using set_advertised_address() before sending the call out. > http://www.opensips.org/Resources/DocsCoreFcn#toc125 > > or use the record_route_preset() function : > http://www.opensips.org/html/docs/modules/devel/rr.html#id228590 > > Regards, > Bogdan > > Daniel Goepp wrote: > > After further investigation, this only updating the Via header, but > > the RR is remaining untouched: > > > > Record-Route: <sip:10.250.7.164;lr=on> > > > > Ideas about how I might get this field updated correctly? > > > > Thanks > > > > -dg > > > > On Wed, Oct 7, 2009 at 2:10 PM, Bogdan-Andrei Iancu > > <[email protected]> wrote: > > > >> Hi Daniel, > >> > >> try using the advertise_address and advertise_port to force opensips in > >> using the public IPs of the NAT: > >> http://www.opensips.org/Resources/DocsCoreFcn#toc24 > >> > >> Regards, > >> Bogdan > >> > >> Daniel Goepp wrote: > >> > >>> I am trying to setup OpenSIPs to run behind a firewall, and not > >>> finding much information regarding how to get OpenSIPs to be aware of > >>> the public IP for it's signaling. The firewall is setup with a 1 to 1 > >>> NAT for the public and private IPs, and right now all udp and tcp > >>> traffic is being passed directly through. Has anyone successfully > >>> gotten OpenSIPs setup like this? If so, can you please provide any > >>> information on how it was setup. > >>> > >>> TIA > >>> > >>> -dg > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> [email protected] > >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >>> > >> _______________________________________________ > >> Users mailing list > >> [email protected] > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > >> > > > > _______________________________________________ > > Users mailing list > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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