Really weird but I unchecked the "Media Termination Point Required" box and now it works. Is this going to get me into trouble down the road?<br><br><div class="gmail_quote">On Wed, Nov 28, 2012 at 10:41 AM, abbas Wali <span dir="ltr"><<a href="mailto:abbaseo@gmail.com" target="_blank">abbaseo@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I had some issues when i created my first SIP trunk between CUCM 8.5 and CUCM 6. after some testing I found out that<br>
<p class="MsoNormal">the Route list for that particular Route pattern <span style="color:red">must </span>be registered with the CM who is participating in
the SIP signalling for the trunk on the remote CM. <br></p><p class="MsoNormal"><br></p><p class="MsoNormal">hope that helps.<br></p>
<br><br><div class="gmail_quote">On 28 November 2012 02:06, Erick Wellnitz <span dir="ltr"><<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Reading the SRND for both 6 and 8 I think I misconfigured the intercluster trunk. I'm going to simplify my configuration and use a single CUCM on each side to begin with. It is a temporary setup so I'm not concerned with redundancy.<div>
<div><div>
<br></div><div><br><br>On Tuesday, November 27, 2012, Peter Slow wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">the thing that's happening to you is not a reliability problem, calls aren't ever supposed to drop - they drop when they're really "dropping" because of network outages, usually.<br>
<br>what devices are you calling from and to? where are THEY? (physically and logically)<br>
<br>Pick one call flow, the simplest one you can that still exhibits your problem, then troubleshoot that specifically. (and tell us what it is please =)<br><br>Until _both_ of your sides are on 7-or-8-dot-somethign my personal recommendation is to use h.323.<br>
<br>let us know what you decide to do, and more about the call flow (devices on either side, proximity of the CUCMs to each other and # of CMs in the cluster etc). It's difficult to provide more than smart-assery without those things ;)<br>
<br>-Peter<br><br><br><br><br><br><div class="gmail_quote">On Tue, Nov 27, 2012 at 8:46 PM, Erick Wellnitz <span dir="ltr"><<a>ewellnitzvoip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">No firewall. I wish it was that simple.<div><div><br><br>On Tuesday, November 27, 2012, Peter Slow wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">remove the firewall that's in between your two clusters?<br>
<br><div class="gmail_quote">On Tue, Nov 27, 2012 at 8:16 PM, Erick <span dir="ltr"><<a>ewellnitzvoip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I'm migrating sites from a 6.1 cucm cluster to an 8.6 cluster.<br>
<br>
I have set up both SIP and an intercluster trunk with similar results. After a random period of time you get a fast busy calling between the clusters.<br>
<br>
Anyone have any tips for settinv this up to be reliable?<br>
<br>
Thanks!<br>
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<br></blockquote></div><span class="HOEnZb"><font color="#888888"><br><br clear="all"><br>-- <br><font style="font-family:comic sans ms,sans-serif;color:rgb(51,102,255)" size="4"><span style="color:rgb(0,0,153)">@bbas.. </span><br>
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