<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 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 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 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"><div><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>
</blockquote>
<br>
</div>
</blockquote></div><br></div>