<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(&quot;INVITE&quot;)) {<br>setflag(1); # do accouting<br>if (uri=~&quot;sip:[0-9]{10,<a href="mailto:11%7D@192.168.7.80">11}@192.168.7.80</a>&quot;)<br>{<br>xlog(&quot;*********CALL WILL GO HERE VOIP INOVATIONS********&quot;);<br>

xlog(&quot;*****************GOING TO ROUTE @6****************&quot;);<br>route(6);<br>}<br><br>}<br><br>route[6] {<br>rewritehost(&quot;67.37.xx.35:5060&quot;); # Provider IP<br>xlog(&quot;************** new ruri=&lt;$ru&gt;, dst=&lt;$du&gt;***********\n&quot;);<br>

xlog(&quot;***********$ru**************\n&quot;);<br>xlog(&quot;*********CALL WILL GO TO VOIP @@@@@@@@@OUT BOUND @@@@@@@********&quot;);<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 href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>

  
    
  
  <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 class="im">
    <pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a 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 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, &quot;Bogdan-Andrei
        Iancu&quot; &lt;<a 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 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(&quot;INVITE&quot;)) {<br>
                    setflag(1); <br>
                    <br>
                    if (uri=~<a>&quot;sip:[0-9]{10,11}@192.168.XX.XX&quot;</a>) 
                    # Asterisk server<br>
                    {<br>
                    xlog(&quot;*********CALL WILL GO HERE VOIP
                    PROVIDER********&quot;);<br>
                    xlog(&quot;*****************GOING TO ROUTE
                    @6****************&quot;);<br>
                    route(6);<br>
                    }<br>
                    <br>
                    }<br>
                    <br>
                    route[6] {<br>
                    <br>
                    rewritehostport(&quot;64.XX.XX.XX:5060&quot;); # VOIP Provider
                    IP Address<br>
                    xlog(&quot;*********CALL WILL GO TO VOIP GATEWAY
                    @@@@@@OUT********&quot;);<br>
                    t_relay();<br>
                    exit;<br>
                    }<br>
                    <br>
                    <br>
                  </div>
                  Thanks<br>
                </div>
                Jagan<br>
              </div>
              <pre><fieldset></fieldset>
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</pre>
            </blockquote>
          </div>
        </blockquote>
      </div>
    </blockquote>
  </div></div></div>

</blockquote></div><br></div>