<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>I'm sure it does...that's what I meant by the re-education part. It doesn't look like it will be a simple drop-in replacement. What ever is, eh?<br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>"Bad grammar makes me [sic]" - Tshirt<br><br><br>----- Original Message -----<br>From: "Mark Holloway" <mh@markholloway.com><br>To: "Lelio Fulgenzi" <lelio@uoguelph.ca><br>Cc: "Chris Ward (chrward)" <chrward@cisco.com>, "Cisco VoIPoE List" <cisco-voip@puck.nether.net><br>Sent: Monday, November 23, 2009 7:20:51 PM GMT -05:00 US/Canada Eastern<br>Subject: Re: [cisco-voip] SIP Trunk Redundancy<br><br><link href="/zimbra/css/msgview.css?v=081117021119" rel="stylesheet">SIP Trunking works great when designed and deployed correctly. <div><div><br><div><div>On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:</div><br class="Apple-interchange-newline"><blockquote><span class="Apple-style-span" style="border-collapse: separate; font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;"><div><div style="font-family: Verdana; font-size: 10pt; color: rgb(0, 0, 0);">Every day, I think to myself, man, SIP isn't all it's cracked up to be......<br><br>OK, not _every_ day, but when I read posts like this I do.<br><br>Seems like there will be some "re-edumacating" necessary when moving to SIP.<br><br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>"Bad grammar makes me [sic]" - Tshirt<br><br><br>----- Original Message -----<br>From: "Chris Ward (chrward)" <<a href="mailto:chrward@cisco.com" style="color: blue; text-decoration: underline;" target="_blank">chrward@cisco.com</a>><br>To: "Ted Nugent" <<a href="mailto:tednugent73@gmail.com" style="color: blue; text-decoration: underline;" target="_blank">tednugent73@gmail.com</a>>, "Cisco VoIPoE List" <<a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline;" target="_blank">cisco-voip@puck.nether.net</a>><br>Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern<br>Subject: Re: [cisco-voip] SIP Trunk Redundancy<br><br><div class="Section1"><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);">You would need to look at the traces to verify, but it may just be the time it takes to failover. You probably need to mess with the SIP profiles and timers to get the trunks to failover in a timely manner. I think by default it may take 15+ seconds (depends on # of retires and time between retries) for a SIP trunk call to failover to the next member of a route group.</span></div><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);"> </span></p><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);">-Chris</span></div><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);"> </span></p><div style="border-style: solid none none; border-width: 1pt medium medium; padding: 3pt 0in 0in;"><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><b><span style="font-size: 10pt; font-family: Tahoma,sans-serif;">From:</span></b><span style="font-size: 10pt; font-family: Tahoma,sans-serif;"><span class="Apple-converted-space"> </span><a href="mailto:cisco-voip-bounces@puck.nether.net" style="color: blue; text-decoration: underline;" target="_blank">cisco-voip-bounces@puck.nether.net</a><span class="Apple-converted-space"> </span>[mailto:cisco-voip-bounces@puck.nether.net]<span class="Apple-converted-space"> </span><b>On Behalf Of<span class="Apple-converted-space"> </span></b>Ted Nugent<br><b>Sent:</b><span class="Apple-converted-space"> </span>Monday, November 23, 2009 2:14 PM<br><b>To:</b><span class="Apple-converted-space"> </span>Cisco VoIPoE List<br><b>Subject:</b><span class="Apple-converted-space"> </span>[cisco-voip] SIP Trunk Redundancy</span></div></div><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"> </p><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span class="apple-style-span">I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.</span></div><div><div><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"> </p></div><div><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><br><br><span class="apple-style-span"></span></div></div></div></div><br>_______________________________________________ cisco-voip mailing list<span class="Apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline;" target="_blank">cisco-voip@puck.nether.net</a><span class="Apple-converted-space"> </span><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" style="color: blue; text-decoration: underline;" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a></div>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline;" target="_blank">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" style="color: blue; text-decoration: underline;" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br></div></span></blockquote></div><br></div></div></div></body></html>