<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    <tt>Ok, then the IPs in SDP cannot ensure the RTP connectivity (some
      private IPs ?) - check the IPs on each SDP and be sure the end
      point can use them to route to the other party.<br>
      <br>
      Regards,<br>
    </tt>
    <pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
    <br>
    On 04/08/2013 04:57 PM, Jagadish Thoutam wrote:
    <blockquote
cite="mid:CAHqn9HEn+8Hq5mGtx8R-4xsrTKAsugtHVyF+0mGLUcQkXO3x6Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>
          <div>
            <div>HI Bogdan,<br>
              <br>
            </div>
             Yes it is.  But No Audio.<br>
            <br>
            <br>
            <br>
            <br>
          </div>
          Thanks<br>
          <br>
        </div>
        Jagadish</div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On 8 April 2013 08:20, Bogdan-Andrei
          Iancu <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
            0.8ex; border-left: 1px solid rgb(204, 204, 204);
            padding-left: 1ex;">
            <div bgcolor="#ffffff" text="#000000"> <tt>Hello,<br>
                <br>
                this still does not answer to my question - does your
                SIP signaling work ok (for the established call) ?<br>
                <br>
                Regards,<br>
              </tt>
              <div class="im">
                <pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
                <br>
              </div>
              <div>
                <div class="h5"> On 04/05/2013 05:36 PM, Jagadish
                  Thoutam wrote:
                  <blockquote type="cite">
                    <div dir="ltr">
                      <div>
                        <div>
                          <div>Hi Bogdan,<br>
                            <br>
                            <br>
                          </div>
                          here is my setup<br>
                          <br>
                        </div>
                        (X-lite)client----------&gt;Asterisk-----&gt;
                        Opensips(NAT) -----&gt;Gateway.<br>
                        <br>
                        <br>
                      </div>
                      <b>################################## Opensips
                        Config File#####################  </b>                                                 

                      <br>
                      <div><b>route{<br>
                          <br>
                          if (is_method("INVITE")) {<br>
                          setflag(1); # do accouting<br>
                          if (uri=~"sip:[0-9]{10,<a
                            moz-do-not-send="true"
                            href="mailto:11%7D@192.168.7.80"
                            target="_blank">11}@192.168.7.80</a>")<br>
                          {<br>
                          xlog("*********CALL WILL GO HERE VOIP
                          INOVATIONS********");<br>
                          xlog("*****************GOING TO ROUTE
                          @6****************");<br>
                          route(6);<br>
                          }<br>
                          <br>
                          }<br>
                          <br>
                          route[6] {<br>
                          rewritehost("67.37.xx.35:5060"); # Provider IP<br>
                          xlog("************** new ruri=&lt;$ru&gt;,
                          dst=&lt;$du&gt;***********\n");<br>
                          xlog("***********$ru**************\n");<br>
                          xlog("*********CALL WILL GO TO VOIP
                          @@@@@@@@@OUT BOUND @@@@@@@********");<br>
                          t_relay();<br>
                          exit;<br>
                          }<br>
                          <br>
                          <br>
                        </b></div>
                      <div><b>Here is My Trace File see attachment <br>
                          <br>
                          <br>
                        </b></div>
                      <div><b>Thanks<br>
                        </b></div>
                      <div><b>Jagadish<br>
                        </b></div>
                      <b><br>
                          </b></div>
                    <div class="gmail_extra"><br>
                      <br>
                      <div class="gmail_quote">On 5 April 2013 09:34,
                        Bogdan-Andrei Iancu <span dir="ltr">&lt;<a
                            moz-do-not-send="true"
                            href="mailto:bogdan@opensips.org"
                            target="_blank">bogdan@opensips.org</a>&gt;</span>
                        wrote:<br>
                        <blockquote class="gmail_quote" style="margin:
                          0pt 0pt 0pt 0.8ex; border-left: 1px solid
                          rgb(204, 204, 204); padding-left: 1ex;">
                          <div bgcolor="#ffffff" text="#000000"> <tt>So
                              you actually have a media problem. Is one
                              way audio or no-audio at all ?<br>
                              <br>
                              As OpenSIPS is nated and the GW public (I
                              assume), is the signaling working properly
                              (INVITE+200OK+ACK) ?<br>
                              <br>
                              Regards,<br>
                            </tt>
                            <div>
                              <pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
                              <br>
                            </div>
                            <div>
                              <div> On 04/05/2013 04:08 PM, Jagadish
                                Thoutam wrote:
                                <blockquote type="cite">
                                  <p dir="ltr">Yes i can see that, even
                                    call is instiating with opensips and
                                    provider but no voice.</p>
                                  <p dir="ltr">My opensips is behind the
                                    NAT, so is there any issue with nat
                                    settings.</p>
                                  <p dir="ltr">Thanks<br>
                                    jagadish.</p>
                                  <p dir="ltr">sent from samsung S3</p>
                                  <div class="gmail_quote">On 5 Apr 2013
                                    20:04, "Bogdan-Andrei Iancu" &lt;<a
                                      moz-do-not-send="true"
                                      href="mailto:bogdan@opensips.org"
                                      target="_blank">bogdan@opensips.org</a>&gt;

                                    wrote:<br type="attribution">
                                    <blockquote class="gmail_quote"
                                      style="margin: 0pt 0pt 0pt 0.8ex;
                                      border-left: 1px solid rgb(204,
                                      204, 204); padding-left: 1ex;">
                                      <div bgcolor="#ffffff"
                                        text="#000000"> <tt>Hello
                                          Jagadish,<br>
                                          <br>
                                          Using a network tracer
                                          (tcpdump, ngrep, wireshark),
                                          do you see the INVITE going
                                          out (sent out by OpenSIPS)  ?<br>
                                          <br>
                                          Regards,<br>
                                        </tt>
                                        <pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
                                        <br>
                                        On 04/04/2013 02:46 AM, Jagadish
                                        Thoutam wrote:
                                        <blockquote type="cite">
                                          <div dir="ltr">
                                            <div>
                                              <div>
                                                <div>Hi All,<br>
                                                  <br>
                                                </div>
                                                i having issue with URI
                                                routing , when i am
                                                trying with the Voip
                                                Provider IP its Not
                                                Going Through, i have IP
                                                authentication with
                                                Provider<br>
                                              </div>
                                              <div><br>
                                                here is the my script<br>
                                              </div>
                                              <br>
                                              <div>if
                                                (is_method("INVITE")) {<br>
                                                setflag(1); <br>
                                                <br>
                                                if (uri=~<a
                                                  moz-do-not-send="true">"sip:[0-9]{10,11}@192.168.XX.XX"</a>) 


                                                # Asterisk server<br>
                                                {<br>
                                                xlog("*********CALL WILL
                                                GO HERE VOIP
                                                PROVIDER********");<br>
                                                xlog("*****************GOING
                                                TO ROUTE
                                                @6****************");<br>
                                                route(6);<br>
                                                }<br>
                                                <br>
                                                }<br>
                                                <br>
                                                route[6] {<br>
                                                <br>
                                                rewritehostport("64.XX.XX.XX:5060");
                                                # VOIP Provider IP
                                                Address<br>
                                                xlog("*********CALL WILL
                                                GO TO VOIP GATEWAY
                                                @@@@@@OUT********");<br>
                                                t_relay();<br>
                                                exit;<br>
                                                }<br>
                                                <br>
                                                <br>
                                              </div>
                                              Thanks<br>
                                            </div>
                                            Jagan<br>
                                          </div>
                                          <pre><fieldset></fieldset>
_______________________________________________
Users mailing list
<a moz-do-not-send="true" href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a moz-do-not-send="true" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
                                        </blockquote>
                                      </div>
                                    </blockquote>
                                  </div>
                                </blockquote>
                              </div>
                            </div>
                          </div>
                        </blockquote>
                      </div>
                      <br>
                    </div>
                  </blockquote>
                </div>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
    </blockquote>
  </body>
</html>