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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>If it were TCP it might be fast as the TCP connection would time
out much faster. For UDP, let’s start with just reducing the number of
retries. The SIP Profile may not do this as I recall, the SIP profile is used
primarily for SIP end points. There are SIP CCM Service parameters that should
cover this. Perhaps in the service window you can try both.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>-Chris<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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"'> Ted Nugent
[mailto:tednugent73@gmail.com] <br>
<b>Sent:</b> Monday, November 23, 2009 3:20 PM<br>
<b>To:</b> Chris Ward (chrward)<br>
<b>Cc:</b> Cisco VoIPoE List<br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk Redundancy<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I will try changing the invites for sure. <o:p></o:p></p>
<div>
<p class=MsoNormal>I noticed that they are using a custom security profile and
I just found out that the provider is currently only accepting UDP for
outgoing. Apparently they had issues when they first set this up and the only
way they could get it working was to lock in the security profile with outgoing
UDP. Think this might be part of the problem? <o:p></o:p></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p>
<div>
<p class=MsoNormal>On Mon, Nov 23, 2009 at 3:03 PM, Chris Ward (chrward) <<a
href="mailto:chrward@cisco.com">chrward@cisco.com</a>> wrote:<o:p></o:p></p>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>I would set the retries for INVITES down
to 1 or 2. Also, are you using TCP or UDP?</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>-Chris</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span
style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> Ted
Nugent [mailto:<a href="mailto:tednugent73@gmail.com" target="_blank">tednugent73@gmail.com</a>]
<br>
<b>Sent:</b> Monday, November 23, 2009 3:00 PM<br>
<b>To:</b> Chris Ward (chrward)<br>
<b>Cc:</b> Cisco VoIPoE List<br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk Redundancy</span><o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Mark<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Unfortunately
there's not CUBE so no Dialpeers to make changes on, any other ideas short of
going with CUBE?<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Chris<o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I'll
try and pull some traces when we can test again, they have a funky maintenance
window so its hard to test anything except late at night. When they first
noticed this and got us engaged they said the sites primary SIP routers power
supply died and it never rolled to the next SIP Trunk, they were able to
physically reorder the routelist members to get outbound calls working on the
next RG in the RL but it appears to not to be rerouting on its own? Is there an
option I'm not seeing to enable trunk failover or something like that?<o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Here
are the current timers under the profile and what the defaults are set to but
if it didn't FO in well over an hour I'm thinking something else might be at
work here. Any thoughts as to which one specifically I'm looking at so I can go
in with a game plan?<o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Timer
Invite Expires (seconds) = 180<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Timer
Register Delta (seconds) = 5<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Timer
Register Expires (seconds) = 3600<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Timer
T1 (msec) = 500<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Timer
T2 (msec) = 4000<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Retry
INVITE = 6 <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Retry
Non-INVITE = 10<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On
Mon, Nov 23, 2009 at 2:21 PM, Chris Ward (chrward) <<a
href="mailto:chrward@cisco.com" target="_blank">chrward@cisco.com</a>>
wrote:<o:p></o:p></p>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>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><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>-Chris</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span
style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> <a
href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]
<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><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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. <o:p></o:p></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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