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<p class="MsoNormal"><span style="color:#1F497D">Have you tried using progress-ind on the pots dialpeers?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">For example:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><i><span style="color:red">dial-peer voice 1 pots<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span style="color:red">&nbsp;progress_ind setup enable 3<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span style="color:red">&nbsp;progress_ind progress enable 8<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span style="color:red">&nbsp;progress_ind connect enable 8<o:p></o:p></span></i></p>
<p class="MsoNormal"><span style="color:#1F497D">&nbsp;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Or you could try the disc_pi_off under the voice-port.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Example:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><i><span style="color:red">voice-port 0/0/0:15<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span style="color:red">disc_pi_off<o:p></o:p></span></i></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal">The disc_pi_off command enables the h323 gateway to disconnect a call when it receives a disconnect message with a progress indicator value (PI).<o:p></o:p></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Hopefully that might help.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">The only other thing I can think of is maybe try the following command:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><i><span style="color:red">voice class h323 1<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span style="color:red">&nbsp;h225 timeout tcp establish 3<o:p></o:p></span></i></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">You will need to apply that to your voip dial-peers that point to your call managers.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><i><span style="color:red">dial-peer voice 2 voip<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span style="color:red">&nbsp;voice-class&nbsp; h323 1<o:p></o:p></span></i></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><a name="secondary">This step is necessary because by default, the H.225 timer that controls redirection to a less preferred dial peer on a &quot;no-response&quot; failure is longer than the ten second timer of the Q.931 Call Proceeding timer. When
 a call comes in to an H.323 gateway through an ISDN trunk and gets forwarded to an inoperative Cisco CallManager, the router waits for forty seconds before it attempts to use a dial peer with a lower preference, or before it clears the call. By the time this
 occurs, the ISDN Q.931 that signals on the H.323 gateway has already sent an ISDN Q.931 CALL DISCONNECT to the ISDN switch. The router provides an ISDN clearing code of 0x8066&#8212;&quot;recovery on timer expiry.&quot;</a><o:p></o:p></p>
<p class="MsoNormal">It is not possible for the H.323 gateway to reset the Call Proceeding timer because it attempts to use a different dial-peer. Therefore, the H.323 gateway must switch peers and complete the call with the help of the secondary Cisco CallManager
 server within the ten seconds allowed by the Q.931 (Incoming Call Proceeding) timer.<o:p></o:p></p>
<p class="MsoNormal">When the H.225 timer is set to three seconds, the router attempts a connection to the primary Cisco CallManager server. If it does not receive a response in three seconds, it falls back to the secondary Cisco CallManager server.<o:p></o:p></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
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<p class="MsoNormal"><b><i><span style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;
color:#003478">Karen Cheng</span></i></b><span style="font-size:12.0pt;
font-family:&quot;Times New Roman&quot;,&quot;serif&quot;;color:#1F497D"><o:p></o:p></span></p>
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<p class="MsoNormal"><i><span style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;
color:#003478">Voice Network Engineer</span></i><span style="font-size:12.0pt;
font-family:&quot;Times New Roman&quot;,&quot;serif&quot;;color:#1F497D"><o:p></o:p></span></p>
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<p class="MsoNormal"><i><span style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;
color:#003478"><a href="http://www.racq.com.au"></a><o:p></o:p></span></i></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;font-family:&quot;Times New Roman&quot;,&quot;serif&quot;;
color:#1F497D">&nbsp;<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;font-family:&quot;Times New Roman&quot;,&quot;serif&quot;;
color:#1F497D">&nbsp;<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;font-family:&quot;Times New Roman&quot;,&quot;serif&quot;;
color:#1F497D">&nbsp;<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;font-family:&quot;Times New Roman&quot;,&quot;serif&quot;;
color:#1F497D">&nbsp;</span><span style="color:#1F497D"><o:p></o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:
&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span lang="EN-US" style="font-size:10.0pt;
font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net]
<b>On Behalf Of </b>O'Brien, Neil<br>
<b>Sent:</b> Tuesday, 9 February 2010 2:39 AM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] Ringback instead of busy/disconnect<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p>&nbsp;</o:p></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;">Hi,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;">I posted about this while back but I thought it was an isolated incident relating to the provider.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;">Basically, from an external phone, if I dial a DDI number that doesn&#8217;t have an extension set up in call manager, I hear ringback for about 20 seconds before
 I get the busy/disconnect tone.&nbsp; As I said, I thought this was an isolated incident on one particular site relating to a BRI but tests have show it actual happens on another site (same CM cluster) for a PRI.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;">I&#8217;m guessing this is might be a H323 signalling issue (all gateways are H323) or Call Manager issue but I don&#8217;t know what to even look for.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;">Any help would be appreciated.&nbsp; I will be logging a TAC case for it but I thought I&#8217;d post anyway in case it&#8217;s a known (obviously not to me) issue.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;">Thanks,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;">Neil<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-IE" style="font-size:10.0pt;font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
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