<div dir="ltr">well, probably you softphone/ip phone, is using some kind of stun or other kind of nat features, so, nothing come to be detected, this can happen, so, if you will be ever using nat, you can force the rtpproxy without nat detection, this will solve your problem, if you read the documentation (<a href="http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854">http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854</a> ) you can see that this test are made over rfc1918 or different ip address from via and signalling <br>
<br>The problem is probably the fact that when the call is stablished, the media cannot traverse, you have the correct ip information on sdp but the router does not permit the session to be opened, so, do a test forcing the use of rtpproxy without the nat detection, just force all trafic throught rtpproxy<br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/10/4 Rodrigo Ferreira <span dir="ltr"><<a href="mailto:rsferreira08@gmail.com" target="_blank">rsferreira08@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">I did that Mike ..<div><br></div><div>my "nat_uac_client" isnt passing in any verification ...</div><div><br></div><div>I did this ..</div><div><br></div><div><div> if ( nat_uac_test("1") ) xlog("UAC TEST = 1");</div>
<div> </div><div> if ( nat_uac_test("2") ) xlog("UAC TEST = 2");</div><div> </div><div> if ( nat_uac_test("4") ) xlog("UAC TEST = 4");</div><div><br></div>
<div> if ( nat_uac_test("8") ) xlog("UAC TEST = 8");</div><div><br></div><div> if ( nat_uac_test("16") ) xlog("UAC TEST = 16");<br></div><div><br></div><div> if ( nat_uac_test("32") ) xlog("UAC TEST = 32");</div>
<div><br></div><div> if ( nat_uac_test("64") ) xlog("UAC TEST = 64");</div></div><div><br></div><div>in the beginning of the script, to see what is happening to my NAT, and i got nothing. </div>
</div><div class="gmail_extra"><div class="im"><br clear="all"><div><div dir="ltr"><div><br></div><div><br></div>Atenciosamente.<div>Eng.° Rodrigo Ferreira</div><div>ITIL v3 Certified</div><div><br></div><div><a href="http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901" target="_blank"><img height="19" width="96"></a><br>
</div></div></div>
<br><br></div><div><div class="h5"><div class="gmail_quote">2013/10/4 Mike Tesliuk <span dir="ltr"><<a href="mailto:mike@ultra.net.br" target="_blank">mike@ultra.net.br</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">That howto is just a sample (with a lot of comments) to better understand of nat configuration (over my understand offcourse), so, you can check and compare with your configuration to identify about something missing<br>
<br><br></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/10/4 Rodrigo Ferreira <span dir="ltr"><<a href="mailto:rsferreira08@gmail.com" target="_blank">rsferreira08@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Yes I did Mike,<div><br></div><div>and my SIP messages are ok. </div><div><br></div><div>I will take a look at your tutorial.</div><div><br></div><div>tks</div></div><div class="gmail_extra"><div>
<br clear="all">
<div><div dir="ltr"><div><br></div><div><br></div>Atenciosamente.<div>Eng.° Rodrigo Ferreira</div><div>ITIL v3 Certified</div><div><br></div><div><a href="http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901" target="_blank"><img height="19" width="96"></a><br>
</div></div></div>
<br><br></div><div><div><div class="gmail_quote">2013/10/3 Mike Tesliuk <span dir="ltr"><<a href="mailto:mike@ultra.net.br" target="_blank">mike@ultra.net.br</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div>Did you try to made some debug rodrigo ? maybe some rule is missing on your route script<br><br>i made a tutorial over version 1.9 that you can check<br><br>[portugues] <a href="http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy" target="_blank">http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy</a><br>
</div>[english] <a href="http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English" target="_blank">http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English</a><br>
<br></div>
<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/10/3 Rodrigo Ferreira <span dir="ltr"><<a href="mailto:rsferreira08@gmail.com" target="_blank">rsferreira08@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div>
<div dir="ltr">Hi guys,<div><br></div><div>After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg.</div>
<div><br></div><div>There's anything else that I could take a look at?</div><div><br></div><div>Thanks<br clear="all"><div><div dir="ltr"><div><br></div><div><br></div>Atenciosamente.<div>Eng.° Rodrigo Ferreira</div>
<div>
ITIL v3 Certified</div><div><br></div><div><a href="http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901" target="_blank"><img height="19" width="96"></a><br>
</div></div></div>
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