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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">
<meta content="text/html; charset=windows-1252"
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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>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
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</pre>
</blockquote>
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