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<tt>Hello,<br>
<br>
this still does not answer to my question - does your SIP
signaling work ok (for the established call) ?<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 04/05/2013 05:36 PM, Jagadish Thoutam wrote:
<blockquote
cite="mid:CAHqn9HHrE7K1LUS1z3Vbf3hrDNJN8c+VLBfN4bmA8Pxis7jEWw@mail.gmail.com"
type="cite">
<div dir="ltr">
<div>
<div>
<div>Hi Bogdan,<br>
<br>
<br>
</div>
here is my setup<br>
<br>
</div>
(X-lite)client---------->Asterisk-----> Opensips(NAT)
----->Gateway.<br>
<br>
<br>
</div>
<b>################################## Opensips Config
File##################### </b>
<br>
<div><b>route{<br>
<br>
if (is_method("INVITE")) {<br>
setflag(1); # do accouting<br>
if (uri=~"sip:[0-9]{10,<a moz-do-not-send="true"
href="mailto:11%7D@192.168.7.80">11}@192.168.7.80</a>")<br>
{<br>
xlog("*********CALL WILL GO HERE VOIP INOVATIONS********");<br>
xlog("*****************GOING TO ROUTE @6****************");<br>
route(6);<br>
}<br>
<br>
}<br>
<br>
route[6] {<br>
rewritehost("67.37.xx.35:5060"); # Provider IP<br>
xlog("************** new ruri=<$ru>,
dst=<$du>***********\n");<br>
xlog("***********$ru**************\n");<br>
xlog("*********CALL WILL GO TO VOIP @@@@@@@@@OUT BOUND
@@@@@@@********");<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"><<a moz-do-not-send="true"
href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span>
wrote:<br>
<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>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 moz-do-not-send="true" 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,
"Bogdan-Andrei Iancu" <<a
moz-do-not-send="true"
href="mailto:bogdan@opensips.org"
target="_blank">bogdan@opensips.org</a>>
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 moz-do-not-send="true" 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("INVITE")) {<br>
setflag(1); <br>
<br>
if (uri=~<a moz-do-not-send="true">"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>
<pre><fieldset></fieldset>
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