<p dir="ltr">That did it ! </p>
<p dir="ltr">Created new sip profile, increased the invite timer then increased members no answer duration to 300. The dn's were in a hunt list diverted cfna to the hunt pilot and it loops for your 15 exactly. </p>
<p dir="ltr">Thanks again chaps!</p>
<div class="gmail_quote">On 12 Jun 2014 15:23, "Roy Ramos" <<a href="mailto:irramos@kosmix.com">irramos@kosmix.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">For your internal calls the device originating the calls should have a sip profile with “</span><span style="font-size:8.5pt;font-family:"Verdana","sans-serif"">Timer
Invite Expires (seconds)<img width="8" height="11" src="cid:image001.gif@01CF861E.D33F5E00" alt="Required Field">”
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">higher than 180 seconds.
<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">This settings affects this field on your SIP INVITE<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:Consolas"><br>
INVITE sip: @SIP/2.0<br>
<br>
Via: SIP/2.0/UDP;branch=z9hG4bKfdd88e6c5c5ddd<br>
<br>
From: <sip:>;tag=25120303~a77a1cd4-e106-49f7-8d9b-1ca102a54b6f-108409009<br>
<br>
To: <sip:@><br>
<br>
Date: Tue, 17 Dec 2013 21:50:06 GMT<br>
<br>
Call-ID: 2f9f0080-2b01c70e-d2dbb9-450213ac@<br>
<br>
Supported: timer,resource-priority,replaces<br>
<br>
Min-SE: 1800<br>
<br>
User-Agent: Cisco-CUCM8.6<br>
<br>
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY<br>
<br>
CSeq: 101 INVITE<br>
<br>
Expires: 180 <------------------------------------<br>
<br>
<br>
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> cisco-voip [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>Brian Meade<br>
<b>Sent:</b> Thursday, June 12, 2014 8:55 AM<br>
<b>To:</b> Nick Britt<br>
<b>Cc:</b> Cisco VOIP<br>
<b>Subject:</b> Re: [cisco-voip] No Answer duration - 15 Minutes<u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">Nick,<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">We might be able to get this working internally but I'm not sure your E1 carrier is going to let you keep the call in an alerting state for that long. For the external call, where do you see the disconnect coming from?<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">If you grab CallManager traces for a SIP to SIP call, we can see which timer is being triggered to disconnect the call if it's coming from CUCM. Same thing for your MGCP scenario.<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">Brian<u></u><u></u></p>
</div>
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<p class="MsoNormal"><u></u> <u></u></p>
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</div>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p>
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<p class="MsoNormal">On Thu, Jun 12, 2014 at 5:14 AM, Nick Britt <<a href="mailto:nickolasjbritt@gmail.com" target="_blank">nickolasjbritt@gmail.com</a>> wrote:<u></u><u></u></p>
<div>
<div>
<p class="MsoNormal">Morning/Afternoon,<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">We have a need for calls to ring a shared line (unanswered) for up to 15 minutes.
<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<div>
<p class="MsoNormal">CUCM 9.1.2 SIP enviroment 8945/9951 endpoints.<u></u><u></u></p>
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</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
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<div>
<p class="MsoNormal">We are currently are seeing disconnected on internal (SIP to SIP calls) after 3 minutes and 2:30minutes for external calls (E1 > MD110 > MGCP > CUCM).
<u></u><u></u></p>
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<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">I can see that the unanswered timers on the line can be set to 300 seconds (this would be a good start) but as mentioned above the call is disconnected after 3 minutes regardless of this setting.
<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
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<div>
<p class="MsoNormal">I have tried increasing the SIP INVITE Timers to more than the default of 18000 milli seconds but to no avail.
<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">Is there any other timers that could be modified to resolve this issue?<u></u><u></u></p>
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<div>
<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal">Cheers<span style="color:#888888"><br clear="all">
<br>
<span>-- </span><br>
<span>- Nick</span></span><u></u><u></u></p>
</div>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
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<p class="MsoNormal"><u></u> <u></u></p>
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</blockquote></div>