<div dir="ltr"><div class="gmail_quote">On Wed, Jul 20, 2011 at 9:46 AM, Michael <span dir="ltr"><<a href="mailto:voip.question@gmail.com">voip.question@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div dir="ltr"><div class="gmail_quote"><div class="im">On Wed, Jul 20, 2011 at 9:39 AM, Mark Sayer <span dir="ltr"><<a href="mailto:datapipes@avtb.co.nz" target="_blank">datapipes@avtb.co.nz</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
A "standard" setup would change Asterisk to listen on 5061. Users<br>
would register to OpenSIPS using 5060 and pass the call. via 5061 to<br>
Asterisk for media handling.<br><br></blockquote></div><div>Hi Mark,<br><br>the actual port used for each server doesn't really matter. The question is related to the setup on both applications. Specifically, can an Asterisk peer/extension be configured on the opensips? How would opensips handle both the registration and invites? Is there an example online somewhere?<br>
</div></div></div></blockquote><div> </div></div>Can anyone assist? Is there a guide for that?<br>
<br>
Thanks.</div>