I have a DID from IPKall that is forwarded to my Asterisk box and I have done no special configuration for SIP URI:<br><br>On IPKall you put the IP address of your asterisk box or the hostname then is Sip phone number you put the did number they gave you and thats it.<br>
<br>On your asterisk box you set your peer in sip.conf then in you extensions.conf, whatever you put in your sip phone number, that's what you have to use as the extension.<br><br><br><div class="gmail_quote">On Thu, Aug 20, 2009 at 2:54 PM, randulo <span dir="ltr"><<a href="mailto:spamsucks2005@gmail.com">spamsucks2005@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="im">On Thu, Aug 20, 2009 at 11:15 AM, SIP<<a href="mailto:sip@arcdiv.com">sip@arcdiv.com</a>> wrote:<br>
> IdeaSIP, GizmoProject, IPTel, maybe OnSIP (don't quote me on that one,<br>
> I'm not sure, but someone around has surely used it), etc, etc. There<br>
> are a lot of alternatives about.<br>
<br>
</div>Sorry, I forgot to mention IdeaSIP.com which works great, I've had an<br>
account for years, and of course your own Asterisk box as Gordon<br>
says. You need to be able to accept call via a SIP URI but that's easy<br>
enough to configure.<br>
<br>
We will be talking about this in relation the "Free DID" subject<br>
tomorrow on <a href="http://VUC.me" target="_blank">http://VUC.me</a><br>
<font color="#888888"><br>
/r<br>
</font><div><div></div><div class="h5"><br>
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