<div><br></div><div>Ok, so when [1] talks about Freeswitch acting as intermediary in order to encrypt further communications between an end node SIP device and the main PBX, it means that the FS intermediary node is actually a full SIP node. The phones will need to auth to this and any calls will be routed to trunks via the dial plan.</div>
<div><br></div><div><br></div><div>[1] <a href="http://wiki.freeswitch.org/wiki/SIP_TLS#Hybrid_Encryption">http://wiki.freeswitch.org/wiki/SIP_TLS#Hybrid_Encryption</a></div><br><br><div class="gmail_quote">On Sat, Jan 30, 2010 at 8:50 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div style="word-wrap:break-word">Freeswitch isn't a proxy, and no, we don't provide support for passthrough auth like this. A proxy would, but not sure of any proxy based solution that would do the srtp work for you.<div>
<br></div><div>Mike</div><div><div></div><div class="h5"><div><br><div><div>On Jan 28, 2010, at 9:08 PM, Nicholas Lee wrote:</div><br><blockquote type="cite">Is there a way to do it transparently? The FS proxies will past though the extension creds.<br>
<br><div class="gmail_quote">On Fri, Jan 29, 2010 at 1:52 PM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Then yes you could use FreeSWITCH to augment your Asterisk install and enable encryption from site to site.<br>
<font color="#888888"><br>
/b<br>
</font><div><br>
<br>
> Unfortunately it's not going to cover every situation.<br>
><br>
><br>
> Nicholas<br>
<br>
<br>
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