<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>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)" <chrward@cisco.com><br>To: "Ted Nugent" <tednugent73@gmail.com>, "Cisco VoIPoE List" <cisco-voip@puck.nether.net><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>
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<p class="MsoNormal"><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></p>
<p class="MsoNormal"><span style="font-size: 11pt; font-family: "Calibri","sans-serif"; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; font-family: "Calibri","sans-serif"; color: rgb(31, 73, 125);">-Chris</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; font-family: "Calibri","sans-serif"; color: rgb(31, 73, 125);"> </span></p>
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<p class="MsoNormal"><b><span style="font-size: 10pt; font-family: "Tahoma","sans-serif";">From:</span></b><span style="font-size: 10pt; font-family: "Tahoma","sans-serif";"> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Ted Nugent<br>
<b>Sent:</b> Monday, November 23, 2009 2:14 PM<br>
<b>To:</b> Cisco VoIPoE List<br>
<b>Subject:</b> [cisco-voip] SIP Trunk Redundancy</span></p>
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<p class="MsoNormal"><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></p>
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