<div dir="ltr">Hi Nate,<div><br></div><div>I want to thank you for your support on this issue. The problem got resolved after enabling PRACK on both SIP trunks.</div><div><br></div><div>Abebe</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Jul 24, 2015 at 5:41 PM, Abebe Amare <span dir="ltr"><<a href="mailto:abucho@gmail.com" target="_blank">abucho@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Nate,<div><br></div><div>When you say incoming h323 dial-peer, are you referring to the voip h323 dial-peer from voice gateway to the call manager? I tried to enable the progress_ind commands under the incoming pots dial-peer but it refused.</div><div><br></div><div>regards,</div><div><br></div><div>abebe</div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Jul 24, 2015 at 5:21 PM, NateCCIE <span dir="ltr"><<a href="mailto:nateccie@gmail.com" target="_blank">nateccie@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d">I think incoming h.323 dial-peer.<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"><a href="http://www.cisco.com/c/en/us/support/docs/voice/h323/22983-ringback.html" target="_blank">http://www.cisco.com/c/en/us/support/docs/voice/h323/22983-ringback.html</a><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"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> Abebe Amare [mailto:<a href="mailto:abucho@gmail.com" target="_blank">abucho@gmail.com</a>] <br><b>Sent:</b> Friday, July 24, 2015 7:58 AM</span></p><div><div><br><b>To:</b> NateCCIE<br><b>Cc:</b> cisco voip<br><b>Subject:</b> Re: [cisco-voip] TEHO using SIP trunk ringback issue<u></u><u></u></div></div><p></p><div><div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">Hi Nate,<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Just to confirm, those commands should be applied on the outgoing pots dial-peer?<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div></div><div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">On Fri, Jul 24, 2015 at 4:38 PM, NateCCIE <<a href="mailto:nateccie@gmail.com" target="_blank">nateccie@gmail.com</a>> wrote:<u></u><u></u></p><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in"><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d">That is it. Just do it on both sides.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d">Do you have</span><u></u><u></u></p><p class="MsoNormal" style="background:white"><span style="font-size:8.5pt;font-family:"Courier New";color:black">progress_ind alert enable 8</span><u></u><u></u></p><p class="MsoNormal" style="background:white"><span style="font-size:8.5pt;font-family:"Courier New";color:black">progress_ind progress enable 8</span><u></u><u></u></p><p class="MsoNormal" style="background:white"><span style="font-size:8.5pt;font-family:"Courier New";color:black">progress_ind connect enable 8</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d">on your h.323 dial-peers?</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> Abebe Amare [mailto:<a href="mailto:abucho@gmail.com" target="_blank">abucho@gmail.com</a>] <br><b>Sent:</b> Friday, July 24, 2015 7:18 AM<br><b>To:</b> NateCCIE<br><b>Cc:</b> cisco voip<br><b>Subject:</b> Re: [cisco-voip] TEHO using SIP trunk ringback issue</span><u></u><u></u></p><div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal">Hi Nate,<u></u><u></u></p><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal">On the SIP profile I enabled as follows<u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal"><img border="0" width="474" height="17" src="cid:image001.png@01D0C5E9.ABFC12A0" alt="Inline image 1"><u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal">Is there any other place where I enable PRACK?<u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p></div></div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal">On Fri, Jul 24, 2015 at 3:38 PM, NateCCIE <<a href="mailto:nateccie@gmail.com" target="_blank">nateccie@gmail.com</a>> wrote:<u></u><u></u></p><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt"><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d">Is PRACK enabled on the sip trunk?</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",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>Abebe Amare<br><b>Sent:</b> Friday, July 24, 2015 1:54 AM<br><b>To:</b> cisco voip<br><b>Subject:</b> [cisco-voip] TEHO using SIP trunk ringback issue</span><u></u><u></u></p><div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal">Hi,<u></u><u></u></p><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal">The call flow is like this:<u></u><u></u></p><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal">CUCM cluster 1 ----SIP Trunk---->CUCM cluster 2---H323---->Voice gateway----ISDN PRI--->PSTN<u></u><u></u></p></div></div><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal">The problem is when calling from clus<span style="font-family:"Arial",sans-serif">ter 1 to the PSTN, the CUCM in cluster 2 plays ring back tone whether the final destination i</span>s reachable or not, we don’t hear the PSTN announcements in case the number is wrong or unreachable or…etc. How to remedy this.<u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p></div><div><p class="MsoNormal">thanks in advance<u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p></div></div></div></div></div></div></blockquote></div><p class="MsoNormal"> <u></u><u></u></p></div></div></div></div></div></blockquote></div><p class="MsoNormal"><u></u> <u></u></p></div></div></div></div></div></blockquote></div><br></div>
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