<div dir="ltr">Found this and it works:<div><br></div><div><a href="https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport">https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport</a><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Sep 12, 2014 at 1:31 AM, Rusty Newton <span dir="ltr"><<a href="mailto:rnewton@digium.com" target="_blank">rnewton@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class="">On Wed, Sep 10, 2014 at 2:14 PM, Rizwan H Qureshi<br>
<<a href="mailto:rizwanhasham@gmail.com">rizwanhasham@gmail.com</a>> wrote:<br>
> Hi Everyone,<br>
> How can I create a TLS based sip trunk between two asterisk servers. I have<br>
> been trying to do it but i dont know how to defined the client certificate<br>
> on the asterisk server. Has anyone tried this?<br>
<br>
</span>There is a tutorial for secure calling with TLS and SRTP here:<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial</a><br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
Rusty Newton<br>
Digium, Inc. | Community Support Manager<br>
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direct: +1 256 428 6200<br>
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</font></span></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr"><font color="#888888"><div>Best Ragards</div><div>Rizwan H Qureshi</div><div><br></div>
<div>V: +971 (0) 528272154</div><div><a href="http://linkedin.com/in/rhqureshi" target="_blank">linkedin.com/in/rhqureshi</a></div><div><br></div><div><br></div></font></div>
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