Direct CUBE to CUCM is easy. The following is a snippet from my asterisk lab sip.conf - very simple, no authentication happening, but sets up trunking to both nodes in the lab cluster and dumps inbound calls to a common context for routing. On the CUCM side, it's nothing but a pretty standard SIP trunk config with nothing changed from the defaults.<br>
<br>[cucm9pub]<br>type=friend<br>context=from-callmanager<br>host=192.168.11.50<br>disallow=all<br>allow=ulaw<br>nat=no<br>qualify=yes<br><br>[cucm9sub]<br>type=friend<br>context=from-callmanager<br>host=192.168.11.51<br>
disallow=all<br>allow=ulaw<br>nat=no<br>qualify=yes<br><br><div class="gmail_extra"><br clear="all">Matthew Saskin<br><a href="mailto:msaskin@gmail.com" target="_blank">msaskin@gmail.com</a><br>203-253-9571<br><br>
<br><br><div class="gmail_quote">On Sun, Dec 2, 2012 at 11:39 AM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Realistically you could probably trunk them together without CUBE. CUCM has a lot more capability in altering the inbound/outbound SIP settings that CUBE was used for previously. CUBE would make it potentially easier and still makes the media exchange simpler, but my guess is with Asterisk for 50 phones this design aspect isn't significant.<br>
<br>I haven't seen any guides for either CUCM or CUBE connectivity to Asterisk, but they're probably out there on the interwebs somewhere.<span class="HOEnZb"><font color="#888888"><br><br>-nick</font></span><div class="HOEnZb">
<div class="h5"><br><div class="gmail_extra"><br><br><div class="gmail_quote">
On Sun, Dec 2, 2012 at 10:33 AM, Matthew Saskin <span dir="ltr"><<a href="mailto:msaskin@gmail.com" target="_blank">msaskin@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
This would not be a problem at all. The short answer is you'd end up having dial peers on the CUBE that route (via destination pattern, etc.) specific calls/DID's to the asterisk IP address and specific calls/DID's to the be6k address.<br>
<br>-matthew<br><div class="gmail_extra"><br clear="all">Matthew Saskin<br><a href="mailto:msaskin@gmail.com" target="_blank">msaskin@gmail.com</a><br><a href="tel:203-253-9571" value="+12032539571" target="_blank">203-253-9571</a><br>
<br>
<br><br><div class="gmail_quote"><div><div>On Sun, Dec 2, 2012 at 12:40 AM, Dieter Jansen <span dir="ltr"><<a href="mailto:justtemprament@gmail.com" target="_blank">justtemprament@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
<p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">Hi All,</p><p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">
We are looking to replace an existing Asterisk PBX with about 40 SIP handsets with a CUCM UC solution (BE6K) with about 50 new CP-8945s.</p><p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">
The current installation uses a SIP trunk and we plan to keep using this to connect to the PSTN. We have a 100 DID block for our incoming calls.</p><p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">
As we have comparatively little experience with the Cisco solution we're looking at a soft introduction and will probably have both sets of handsets in place in parallel for a while.</p><p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">
I've got all the edge switches and VLANing working - independent data, voice and legacy voice VLANs/subnets and all required routing between. We're at the point where test phones can place video calls to each other so we now need to get routing and PSTN access happening</p>
<p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">Originally I was thinking I'd set up a CUCM-behind-Asterisk or Asterisk behind-CUCM situation whilst testing but on reading the Cisco documents and books I've noted the suggested use of a CUBE SBC. As we have a spare router that can probably do CUBE I was wondering if I should set that up to register with our VSP.</p>
<p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">In the first instance I'd get the Asterisk box to pass calls to the CUBE and have the CUBE forward all incoming calls to the Asterisk. </p>
<p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">What I'm hoping is that I could then also have the CUCM pass calls to the same CUBE and route a subset of the inward DIDs to the CUCM whilst testing - allowing me to get all the setup sorted whilst easily flipping incoming calls to specific DID between Asterisk and CUCM.</p>
<p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">I've not seen an example of the CUBE being used in this way - is it possible?</p><p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)">
Regards, JustTemprament.</p><p style="margin:0.8em 0px;font-family:Verdana,Arial,Helvetica,sans-serif;font-size:13px;background-color:rgb(238,238,238)"><br></p>
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