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<tt>Hello Arthur,<br>
<br>
The OpenSIPS script allows you to implement whatever logic you
want, so the answer is : yes, you can do that. <br>
<br>
Reuse the part for handling the sequential requests from the
default opensips script and for initial requests (after handling
CANCEL and retransmissions) you can simply do :<br>
$du = <a class="moz-txt-link-rfc2396E" href="sip:asterisk_ip:asterisk_port">"sip:asterisk_ip:asterisk_port"</a>;<br>
t_relay;<br>
<br>
This will send the INVITE to asterisk without changing the RURI at
all.<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 04/17/2013 12:51 PM, Arthur Titeica wrote:
<blockquote cite="mid:1645931.7umPBzGBob@lhart" type="cite">
<pre wrap="">
Hello,
I'm rather new to opensisps.
>From what I read this is not possible but I thought I should ask just to make
sure.
Is there a configuration setup so that opensips doesn't handle
users/extensions but just forwards everything that matches the configured
domains to an asterisk gateway?
Something like <a class="moz-txt-link-abbreviated" href="mailto:any_ext@sip.example.com">any_ext@sip.example.com</a> should be allowed to go to asterisk and
let asterisk to deal with auth.
Thank you all for any input.
</pre>
<pre wrap="">
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</blockquote>
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