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<div class="moz-cite-prefix"><tt>Hey,<br>
<br>
your script does not do any create_dialog() + topology_hiding()
for the initial INVITEs - so the TH will not work for you, the
calls will simply work in normal way.<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>
On 02.09.2014 19:53, Satish Patel wrote:<br>
</div>
<blockquote
cite="mid:CAPgF-frEHfE3sP97sFbuM9+dzYTF_JoUkKTqOe2bjWkBFfBYwA@mail.gmail.com"
type="cite">
<div dir="ltr">
<div>
<div>
<div>Yes, I have send you email in detail about my scenario,
but Yes, you are right, UA will register to opensips and
dial 00123456789, opensips will route that call to
Freeswitch, Freeswitch has dialplan to send that call back
to opensips removing 00 prefix and then opensips will
forward that call to PSTN or outside gateway, Reason i
need this solution so later i can add more and more FS
without edit any PSTN setting, or my outgoing single IP
will be white list. <br>
<br>
</div>
I have solved issue related 482, and my call routing to
outside so that part is working, but again if PSTN callee
hangup phone then I am getting error "404 not here" so in
short it is not handling BYE properly... I have tried
match_dialog() in loose_route() function but didn't help. <br>
<br>
<br>
</div>
Here is my config, could you please take a look and let me
know if anything missing <a moz-do-not-send="true"
href="http://pastebin.com/KL4ZWnM8">http://pastebin.com/KL4ZWnM8</a><br>
<br>
</div>
also do i need to use record_route() function? <br>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Tue, Sep 2, 2014 at 4:41 AM,
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:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div><tt>Hi Satish,<br>
<br>
SO you have the same call going twice through same FS
box ? and when hitting for the second time you get the
482 ? (once again, for the same call)<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>
<div>
<div class="h5"> On 01.09.2014 21:34, Satish Patel
wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
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<div class="h5">
<div dir="ltr">
<div>
<div>I have following setup, UA register to
Opensips and opensips send call to FS
(freeswitch) and again freeswitch send call
back to opensips and then call get outside
routed. in following Senior freeswitch sending
482 Loop detect error, How do i achieve
following scenario? <br>
<br>
</div>
[UA]------[Opensips]------[SIP Provider]<br>
|<br>
|<br>
|<br>
</div>
[FS]<br>
</div>
<br>
<fieldset></fieldset>
<br>
</div>
</div>
<pre>_______________________________________________
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</pre>
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