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    <tt>Hi Fabrizio,<br>
      <br>
      First of all, be sure your script execution gets to the
      do_routing() part. Use the script_trace() function to trace the
      execution of your script:<br>
         
      <a class="moz-txt-link-freetext" href="http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc43">http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc43</a><br>
      <br>
      Best regards,    </tt><br>
    <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>
    <div class="moz-cite-prefix">On 01.06.2015 18:29, Fabrizio Pappolla,
      Brainy Forge S.r.l. wrote:<br>
    </div>
    <blockquote cite="mid:556C7A46.4040802@brainyforge.it" type="cite">
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      Hi.<br>
      I'm using opensips-1.10.<br>
      I have successful configured the uac_registrant module to my VoIP
      carrier, so my OpenSIPS is registered; I've also correctly
      configured some user: 1000 and 1001. I can register to OpenSIPS
      with this extension and they <span id="result_box"
        class="short_text" lang="en"><span></span> <span class="hps">can
          call</span> <span class="hps">each other. If I try to call
          external number I get "SIP 404 - NOT FOUND" error.<br>
          <br>
          How can I make external call? I've tried to configure drouting
          module, without any success.<br>
          <br>
          <i>opensipsctl dr show</i><br>
          <b>dr gateways</b><br>
+----+------+------+-----------------+-------+------------+-------+------------+-------------+<br>
          | id | gwid | type | address         | strip | pri_prefix |
          attrs | probe_mode | description |<br>
+----+------+------+-----------------+-------+------------+-------+------------+-------------+<br>
          |  2 | 1    |    0 | voip.eutelia.it |     0 | 0          |
          1     |          2 | eutelia     |<br>
+----+------+------+-----------------+-------+------------+-------+------------+-------------+<br>
          <b>dr groups</b><br>
          +----+----------+-----------------+---------+-------------+<br>
          | id | username | domain          | groupid | description |<br>
          +----+----------+-----------------+---------+-------------+<br>
          |  1 | 1000     | voip.eutelia.it |       1 |             |<br>
          +----+----------+-----------------+---------+-------------+<br>
          <b>dr carriers</b><br>
          +----+-----------+--------+-------+-------+-------------+<br>
          | id | carrierid | gwlist | flags | attrs | description |<br>
          +----+-----------+--------+-------+-------+-------------+<br>
          |  1 | 1         | 1      |     0 |       | EUTELIA     |<br>
          +----+-----------+--------+-------+-------+-------------+<br>
          <b>dr rules</b><br>
+--------+---------+--------+---------+----------+---------+--------+-------+-------------+<br>
          | ruleid | groupid | prefix | timerec | priority | routeid |
          gwlist | attrs | description |<br>
+--------+---------+--------+---------+----------+---------+--------+-------+-------------+<br>
          |      1 | 1       |     |         |        0 | 1       |
          1,#1   |       |             |<br>
+--------+---------+--------+---------+----------+---------+--------+-------+-------------+<br>
          <br>
          <br>
          <i>opensipsctl dr gw_status</i><br>
          ID:: 1 IP=voip.eutelia.it <b>Enabled=no</b><br>
          <br>
          If i try to enable it, it goes down after few sec<br>
          <br>
          <i>opensipsctl dr carrier_status</i><br>
          ID:: 1 Enabled=yes<br>
          <br>
          Is the dynamic routing way correct? I need to forward all sip
          traffic from a user (that register to OpenSIPS) to its
          associated trunk ( by uac_registrant module ) and opposite.<br>
          Regards.<br>
        </span></span> <br>
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