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<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 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 04/04/2013 02:46 AM, Jagadish Thoutam wrote:
<blockquote
cite="mid:CAHqn9HEBVeaSLVrAMPEG-X9WaubrR_DBkOqh5z7a9RmvNh2KaA@mail.gmail.com"
type="cite">
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<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 class="moz-txt-link-rfc2396E" href="sip:[0-9]{10,11}@192.168.XX.XX">"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>
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</pre>
</blockquote>
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