I did - registration is purely in Kamailio.<div>In Asterisk - I created sip account for Kamailio based on IP address without username and password.</div><div>This way - all calls from Kamailio go to Asterisk without problems.</div>
<div>In Kamailio I allowed calls from Asterisks.</div><div>You do not need realtime in Asterisk, because Kamailio do all registrations perfectly well.</div><div><br><br><div class="gmail_quote">On Wed, Feb 15, 2012 at 9:36 AM, Mark Sayer <span dir="ltr"><<a href="mailto:datapipes@avtb.co.nz">datapipes@avtb.co.nz</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">We've created custom processes to accomplish this. FreePBX may work<br>
but as it will assume a standalone Asterisk setup it may just cause<br>
problems.<br>
<span class="HOEnZb"><font color="#888888"><br>
Mark<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
On Wed, Feb 15, 2012 at 7:18 AM, Greg Mannie <<a href="mailto:greg@latigi.com">greg@latigi.com</a>> wrote:<br>
> Hello<br>
><br>
> Sorry for another newbie question, but eventually with your "greatly<br>
> appreciated" help I will get proficient in this application.<br>
><br>
> After reading much RFC reading and docs for Kamailio I see where the<br>
> benefits of using the sip proxy for registering devices while using asterisk<br>
> for voicemail or ivr etc. has great benefit.<br>
><br>
> I am not finding much on my end user interaction. If I use realtime<br>
> integration and have multi domain use on kamailio, how do I allow the end<br>
> user to configure their own IVR? Is it possible to use modules like the<br>
> call flow control from freepbx and allow users to configure this themselves?<br>
><br>
> Regards,<br>
><br>
> Greg<br>
><br>
><br>
><br>
><br>
><br>
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