Hi Guys,<br><br>I had a look at this service the other day.. havent tried to get it going yet.<br><br>Im planning on buying a small CME box or ISR with CUBE anyway.<br><br>But in the meantime, I did notice you can change the authentication from either username or to IP address.<br>
<br>Would CUCM 7.1 work with the IP address? Or will this still require a register?<br><br>Also, does CUCM 8 add any more capability on the SIP trunk side?<br><br>I do realise CUBE is the way to go in production, but just from a lab scenario really.<br>
<br>Cheers,<br><br>Tim.<br><br><div class="gmail_quote">On Thu, Dec 17, 2009 at 2:16 AM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com">matthnick@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;">
In related news - the UC520/CME was certified with the Skype SIP<br>
trunk, so it's a paved road hopefully. CUBE shares this exact code so<br>
the certification applies (in all practical theory).<br>
<font color="#888888"><br>
-nick<br>
</font><div><div></div><div class="h5"><br>
On Wed, Dec 16, 2009 at 9:23 AM, Joe Martini (joemar2)<br>
<<a href="mailto:joemar2@cisco.com">joemar2@cisco.com</a>> wrote:<br>
> You need a CUBE since Skype requires a REGISTER which CallManager will<br>
> not do.<br>
><br>
> Joe<br>
><br>
> -----Original Message-----<br>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>
> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Beck, Philipp<br>
> Sent: Wednesday, December 16, 2009 9:12 AM<br>
> To: Cisco VOIP Newsletter - <a href="http://puck.nether.net" target="_blank">puck.nether.net</a><br>
> Subject: [cisco-voip] Cisco CUCM 7.1.2 with Skype for SIP Open Beta<br>
><br>
> Hi all,<br>
> I saw that the Skype for SIP is now in Open Beta. Is there anyone how<br>
> has already set this up and it is working? Do I need a CUBE or is it ok<br>
> if I use a standard CallManager SIP Trunk?<br>
><br>
> Thanks for any response.<br>
> Philipp<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br><br>Cheers,<br><br>Tim<br><br><br>Sent from Wollongong, Nsw, Australia