<div>Erik,</div>
<div><br> The only TRUE way to simulate it is to bring down MGCP and truly enter the DEFAULT applicaiton state (H.323 / SRST). If all you're trying to test is the ability to write H.323 dial peers then you can do just that. The problem is that unless you break MGCP, it doesn't go into fallback state from a system perspective. In the MGCP fallback config segment, it refers to Application Alternate DEFAULT (differs from version to version on the syntax). That's the mode you go into after MGCP dies. Once you're there, the MGCP dial peers go down and your
H.323 dial peers take over (including your PRI local association). That's going to be the only real way to test if SRST is going to work. </div>
<div> </div>
<div>MAKE SURE you configure the Applicaiton Alternate DEFAULT. failure to do so will give VERY eratic results (personal experience) and HOURS of lost bliss banging your head against the wall.<br> </div>
<div>my 2 pennies.</div>
<div> </div>
<div><span class="gmail_quote">On 10/23/07, <b class="gmail_sendername">Erik Stillman</b> <<a href="mailto:Erik.Stillman@shearman.com">Erik.Stillman@shearman.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Thanks Andrew, it's exactly what I did.<br><br><br>Anyway to get those dial-peers working with like an FXO?
<br><br>-----Original Message-----<br>From: "Riley, Andrew" <<a href="mailto:Andrew.Riley@getronics.com">Andrew.Riley@getronics.com</a>>@SHSDOMAIN<br>Sent: Tuesday, October 23, 2007 3:29 PM<br>To: Erik Stillman;
<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>Subject: RE: [cisco-voip] SRST<br><br>Put an ACL in the network to force the one phone to register to SRST.<br><br>If you are doing MGCP, then the ISDN interface is under the control of
<br>CCM and won't work with the dial peers.<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 Erik Stillman<br>Sent: Tuesday, 23 October 2007 7:11 AM<br>To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>Subject: [cisco-voip] SRST<br>
<br>Hey guys, is it possible to test one single phone on SRST while the rest<br>of the site is registered to CallManager?<br><br>I can get the test registered with the gateway, but it won't dial out.<br>Does this have something got do with the PRI being MGCP controlled?
<br><br>My goal is to do some testing without having to take the site down.<br><br>All input appreciated!<br><br>Many thanks,<br>Erik<br><br><br><br>-----Original Message-----<br>From: cisco-voip-bounces@puck.nether.net@SHSDOMAIN On Behalf Of
<br><a href="mailto:cisco-voip-request@puck.nether.net">cisco-voip-request@puck.nether.net</a><br>Sent: Monday, October 22, 2007 4:13 PM<br>To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
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