Robert thanks for weighing in.<div><br></div><div>So, you are saying that FreeSwitch on it's own can tackle issues like this without the need of OpenSIPs? Can you elaborate please?</div><div><br></div><div>Thanks<br><br>
<div class="gmail_quote">On Sat, Jul 23, 2011 at 12:17 AM, Robert-iPhone <span dir="ltr"><<a href="mailto:rhuddleston@gmail.com">rhuddleston@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
I like to put mine on 3389<br>
<br>
hahaha just kidding.<br>
<br>
Personally I'm starting to convert to FreeSwitch - oops I had to say it.<br>
<br>
Security can be difficult and there are some good SBCs out there - just begs investment in technology - OH and bright staff<br>
<br>
<br>
Sent from my iPhone<br>
<div><div></div><div class="h5"><br>
On Jul 23, 2011, at 12:09 AM, Steve Edwards <<a href="http://asterisk.org" target="_blank">asterisk.org</a>@<a href="http://sedwards.com" target="_blank">sedwards.com</a>> wrote:<br>
<br>
> On Fri, 22 Jul 2011, Bruce B wrote:<br>
><br>
>> 1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS actually give me the full capability to the SIP stack to do the sort of thing I was asking for? And this can run on the same server as Asterisk is running?<br>
><br>
> Configure OpenSIPS to listen to 5060 and Asterisk to listen to 5061.<br>
><br>
> --<br>
> Thanks in advance,<br>
> -------------------------------------------------------------------------<br>
> Steve Edwards <a href="mailto:sedwards@sedwards.com">sedwards@sedwards.com</a> Voice: <a href="tel:%2B1-760-468-3867" value="+17604683867">+1-760-468-3867</a> PST<br>
> Newline Fax: <a href="tel:%2B1-760-731-3000" value="+17607313000">+1-760-731-3000</a><br>
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