<div dir="ltr">Hi Bogdan,<div>Thank you very much for your reply, so can you help me with below scenario,<br><br>-> i will be doing all sip registrations on Opensips,which is fine, but i wanted all type of calls (like local, trunk pstn) should be handled through media servers like Asterisk/freeswitch, is that possible.? </div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Jul 11, 2016 at 3:58 PM, 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">
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<tt>Hi,<br>
<br>
The features are implement by doing OpenSIPS scripting (based on
how such services should work in SIP). There is no explicit
documentation on how to implement something like this.<br>
<br>
Regards,<br>
</tt><span class="">
<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>
</span><div><div class="h5"><div>On 11.07.2016 13:14, Siju Nair wrote:<br>
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<div class="gmail_quote">Hi Sir<br>
<br>
Thank you very much sir for your quick reply.. Can you please
sent me a link of documents to achieve those features, since
am unable to find one. <br>
<br>
Thanks,<br>
Siju Nair<br>
<br>
Sent from my iPhone<br>
<br>
> On 07-Jul-2016, at 3:16 PM, Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org" target="_blank"></a><a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>
wrote:<br>
><br>
> Bogdan-Andrei Iancu<br>
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