Thanks for trying. In addition, here's the summary of a packet capture of me trying to call myself:<br><br><a href="http://pastebin.com/raw.php?i=3iczgtXb">http://pastebin.com/raw.php?i=3iczgtXb</a><br><pre> X-Lite = x.x.x.x
OpenSIPS = y.y.y.y
FreeSWITCH = z.z.z.z
TCP 1 0.000000 x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE sip:1111@y.y.y.y;transport=tcp, with session description
TCP 2 0.000042 y.y.y.y -> x.x.x.x TCP sip > 59498 [ACK] Seq=1 Ack=982 Win=864 Len=0
TCP 3 0.000800 y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try
TCP 4 0.000948 y.y.y.y -> z.z.z.z TCP 41695 > sip [SYN] Seq=0 Win=5840 Len=0 MSS=1460 TSV=31556561 TSER=0 WS=6
TCP 5 0.001335 z.z.z.z -> y.y.y.y TCP sip > 41695 [SYN, ACK] Seq=0 Ack=1 Win=14480 Len=0 MSS=1460 TSV=31542864 TSER=31556561 WS=6
TCP 6 0.001385 y.y.y.y -> z.z.z.z TCP 41695 > sip [ACK] Seq=1 Ack=1 Win=5888 Len=0 TSV=31556562 TSER=31542864
TCP 7 0.001696 y.y.y.y -> z.z.z.z SIP/SDP Request: INVITE sip:1111@z.z.z.z;transport=tcp, with session description
TCP 8 0.001855 z.z.z.z -> y.y.y.y TCP sip > 41695 [ACK] Seq=1 Ack=1233 Win=16960 Len=0 TSV=31542865 TSER=31556562
TCP 9 0.002719 z.z.z.z -> y.y.y.y SIP Status: 100 Trying
TCP 10 0.002732 y.y.y.y -> z.z.z.z TCP 41695 > sip [ACK] Seq=1233 Ack=491 Win=6912 Len=0 TSV=31556563 TSER=31542865
TCP 11 0.215286 x.x.x.x -> y.y.y.y TCP 59498 > sip [ACK] Seq=982 Ack=325 Win=4150 Len=0
TCP 12 0.712565 z.z.z.z -> y.y.y.y TCP 48269 > sip [SYN] Seq=0 Win=14600 Len=0 MSS=1460 TSV=31543575 TSER=0 WS=6
TCP 13 0.712628 y.y.y.y -> z.z.z.z TCP sip > 48269 [SYN, ACK] Seq=0 Ack=1 Win=5792 Len=0 MSS=1460 TSV=31557273 TSER=31543575 WS=6
TCP 14 0.712944 z.z.z.z -> y.y.y.y TCP 48269 > sip [ACK] Seq=1 Ack=1 Win=14656 Len=0 TSV=31543576 TSER=31557273
TCP 15 0.712997 z.z.z.z -> y.y.y.y SIP/SDP Request: INVITE sip:1111@x.x.x.x:59498;rinstance=2e3da39d713a676d;transport=TCP, with session description
TCP 16 0.713009 y.y.y.y -> z.z.z.z TCP sip > 48269 [ACK] Seq=1 Ack=1383 Win=8704 Len=0 TSV=31557273 TSER=31543576
TCP 17 0.714476 y.y.y.y -> z.z.z.z SIP Status: 100 Giving a try
TCP 18 0.714635 z.z.z.z -> y.y.y.y TCP 48269 > sip [ACK] Seq=1383 Ack=315 Win=15680 Len=0 TSV=31543577 TSER=31557275
UDP 19 0.714713 y.y.y.y -> x.x.x.x SIP/SDP Request: INVITE sip:1111@x.x.x.x:59498;rinstance=2e3da39d713a676d;transport=TCP, with session description
UDP 20 1.217530 y.y.y.y -> x.x.x.x SIP/SDP Request: INVITE sip:1111@x.x.x.x:59498;rinstance=2e3da39d713a676d;transport=TCP, with session description
UDP 21 2.219539 y.y.y.y -> x.x.x.x SIP/SDP Request: INVITE sip:1111@x.x.x.x:59498;rinstance=2e3da39d713a676d;transport=TCP, with session description</pre>You can see that OpenSIPS is trying to send the request as UDP even though it originally came from FreeSWITCH as TCP.<br>
<br>Regards,<br>William<br><br><br><br><div class="gmail_quote">On Tue, Jan 24, 2012 at 4:28 PM, <span dir="ltr"><<a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Well hell. Thats all I know to do just from looking at the documentation. Hopefully one of the other guys can help when they wake up in their respective timezones.
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<br>Sorry I couldn't help.
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<br>On , William Edwards <<a href="mailto:shadowapex@gmail.com" target="_blank">shadowapex@gmail.com</a>> wrote:
<br>> Hey Duane,
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<br>> Yes, I tried using force_send_socket, but it still sends the request as UDP instead of the original TCP. I noticed in the documentation it says that if the protocol doesn't match, the closest socket of the same protocol is used:
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<br>> "If the protocol doesn't match (e.g. UDP message "forced" to a TCP socket) the closest socket of the same protocol is used.
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<br>> Regards,
<br>> William
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<br>> On Tue, Jan 24, 2012 at 4:14 PM, <a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>> wrote:
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<br>> What about force_send_socket
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<br>> <a href="http://www.opensips.org/Resources/DocsCoreFcn#toc111" target="_blank">http://www.opensips.org/Resources/DocsCoreFcn#toc111</a>
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<br>> Have you tried that yet.
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<br>> force_send_socket(udp:<a href="http://192.168.1.10:5060" target="_blank">192.168.1.10:5060</a>);
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<br>> On , William Edwards <a href="mailto:shadowapex@gmail.com" target="_blank">shadowapex@gmail.com</a>> wrote:
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<br>> > In my attempt to resolve this problem I have also already come across that article. Unfortunately the t_relay function does not support variables as parameters, so I am not able to put in the full uri (e.g. tcp:70.232.x.x:55085) in the function. From this article (<a href="http://openser.org/pipermail/users/2010-March/011638.html" target="_blank">http://openser.org/pipermail/users/2010-March/011638.html</a>), it was suggested to rewrite the $du instead, however I don't believe it is possible to specify the protocol in the $du.
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<br>> > On Tue, Jan 24, 2012 at 3:12 PM, <a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>> wrote:
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<br>> > I did a search for "tcp" on Nabble's OpenSIPS section and just found this
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<br>> > I would say you need to use t_relay() with parameters.
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<br>> > Hope that helps.
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<br>> > On , William Edwards <a href="mailto:shadowapex@gmail.com" target="_blank">shadowapex@gmail.com</a>> wrote:
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<br>> > > Thanks for your reply. I've looked into $fs, but I haven't been able to find any concise documentation about how I can use it to force the protocol to TCP. It's really unclear as to why OpenSIPS would be rewritting the TCP message to UDP in the first place. Is there no way to configure OpenSIPS to relay the message using the same protocol that it was received?
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<br>> > > On Tue, Jan 24, 2012 at 10:26 AM, <a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>> wrote:
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<br>> > > Perhaps you need this somewhere in your script
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<br>> > > $fs Forced socket: <a href="http://www.opensips.org/Resources/DocsCoreVar16#toc38" target="_blank">http://www.opensips.org/Resources/DocsCoreVar16#toc38</a>
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<br>> > > On , William Edwards <a href="mailto:shadowapex@gmail.com" target="_blank">shadowapex@gmail.com</a>> wrote:
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<br>> > > > Hi,
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<br>> > > > We currently have two Freeswitch boxes behind OpenSIPS which
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<br>> > > > is acting as a SIP proxy/load balancer. Right now we're facing a problem
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<br>> > > > Freeswitch servers from TCP to UDP. This, of course, is causing a
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<br>> > > > This is an overview of what is happening:
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<br>> > > > Freeswitch -TCP-> OpenSIPS -UDP-> Client UA (listening on TCP)
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<br>> > > > We would like OpenSIPS to retain the original protocol that the message was received. Is there any way this is possible?
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<br>> > > > My current OpenSIPS configuration can be found here:
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<br>> > > > <a href="http://pastebin.com/DSU1f2VS" target="_blank">http://pastebin.com/DSU1f2VS</a>
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<br>> > > > Regards,
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<br>> > > _______________________________________________
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<br>> > > Users mailing list
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<br>> > > <a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
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<br>> > _______________________________________________
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<br>> > Users mailing list
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<br>> > <a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
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<br>> _______________________________________________
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<br>> Users mailing list
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<br>> <a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
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<br>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
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<br></blockquote></div><br>