<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Cisco Press has a SIP Trunking book coming out in January 2010.<div><br></div><div>It's really up to the ITSP to test and certify each IP PBX for their SIP network. It's a painful process but if done correctly, it will work as expected and customers are happy. </div><div><br></div><div><br></div><div><div><div>On Nov 23, 2009, at 5:49 PM, Thomas LeMay wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><o:smarttagtype namespaceuri="urn:schemas-microsoft-com:office:smarttags" name="country-region">
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<div class="Section1"><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy">Hi, Mark,<o:p></o:p></span></font></p><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy"><o:p> </o:p></span></font></p><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy">Can you suggest any best business
practices for the design and deployment of SIP trunking? Can you point me
to any good books on SIP design and deployment?<o:p></o:p></span></font></p><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy"><o:p> </o:p></span></font></p><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy">Thanks,<o:p></o:p></span></font></p><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy"><o:p> </o:p></span></font></p><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy">Tom<o:p></o:p></span></font></p><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy"><o:p> </o:p></span></font></p>
<div>
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<hr size="2" width="100%" align="center" tabindex="-1">
</span></font></div><p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size:10.0pt;
font-family:Tahoma;font-weight:bold">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma"> <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style="font-weight:bold">On
Behalf Of </span></b>Mark Holloway<br>
<b><span style="font-weight:bold">Sent:</span></b> Monday, November 23, 2009
7:21 PM<br>
<b><span style="font-weight:bold">To:</span></b> Lelio Fulgenzi<br>
<b><span style="font-weight:bold">Cc:</span></b> Cisco VoIPoE List<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [cisco-voip] SIP
Trunk Redundancy</span></font><o:p></o:p></p>
</div><p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:
12.0pt"><o:p> </o:p></span></font></p><p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:
12.0pt">SIP Trunking works great when designed and deployed correctly. <o:p></o:p></span></font></p>
<div>
<div><p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:
12.0pt"><o:p> </o:p></span></font></p>
<div>
<div><p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:
12.0pt">On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:<o:p></o:p></span></font></p>
</div><p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:
12.0pt"><br>
<br>
<o:p></o:p></span></font></p>
<div>
<div><p class="MsoNormal" style="margin-bottom:12.0pt"><font size="2" color="black" face="Verdana"><span style="font-size:10.0pt;font-family:Verdana;color:black"><span style="orphans: 2;widows: 2;-webkit-border-horizontal-spacing: 0px;-webkit-border-vertical-spacing: 0px;
-webkit-text-decorations-in-effect: none;-webkit-text-size-adjust: auto;
-webkit-text-stroke-width: 0px;word-spacing:0px">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) * <st1:placetype w:st="on">University</st1:placetype> of <st1:placename w:st="on">Guelph</st1:placename> * <st1:place w:st="on"><st1:city w:st="on">Guelph</st1:city>,
<st1:state w:st="on">Ontario</st1:state> <st1:postalcode w:st="on">N1G 2W1</st1:postalcode></st1:place><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">chrward@cisco.com</a>><br>
To: "Ted Nugent" <<a href="mailto:tednugent73@gmail.com">tednugent73@gmail.com</a>>,
"Cisco VoIPoE List" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 <st1:country-region w:st="on">US</st1:country-region>/<st1:country-region w:st="on"><st1:place w:st="on">Canada</st1:place></st1:country-region> Eastern<br>
Subject: Re: [cisco-voip] SIP Trunk Redundancy<o:p></o:p></span></span></font></p>
<div>
<div><p class="MsoNormal"><font size="2" color="#1f497d" face="Calibri"><span style="font-size:11.0pt;font-family:Calibri;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></font><font color="black"><span style="color:black"><o:p></o:p></span></font></p>
</div><p class="MsoNormal"><font size="2" color="#1f497d" face="Calibri"><span style="font-size:11.0pt;font-family:Calibri;color:#1F497D"> </span></font><font color="black"><span style="color:black"><o:p></o:p></span></font></p>
<div><p class="MsoNormal"><font size="2" color="#1f497d" face="Calibri"><span style="font-size:11.0pt;font-family:Calibri;color:#1F497D">-Chris</span></font><font color="black"><span style="color:black"><o:p></o:p></span></font></p>
</div><p class="MsoNormal"><font size="2" color="#1f497d" face="Calibri"><span style="font-size:11.0pt;font-family:Calibri;color:#1F497D"> </span></font><font color="black"><span style="color:black"><o:p></o:p></span></font></p>
<div style="border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in">
<div><p class="MsoNormal"><b><font size="2" color="black" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;color:black;font-weight:bold">From:</span></font></b><span class="apple-converted-space"><font size="2" color="black" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;color:black"> </span></font></span><font size="2" color="black" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;
color:black"><a href="mailto:cisco-voip-bounces@puck.nether.net">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><span style="font-weight:bold">On
Behalf Of<span class="apple-converted-space"> </span></span></b>Ted Nugent<br>
<b><span style="font-weight:bold">Sent:</span></b><span class="apple-converted-space"> </span>Monday, November 23, 2009 2:14 PM<br>
<b><span style="font-weight:bold">To:</span></b><span class="apple-converted-space"> </span>Cisco VoIPoE List<br>
<b><span style="font-weight:bold">Subject:</span></b><span class="apple-converted-space"> </span>[cisco-voip] SIP Trunk Redundancy</span></font><font color="black"><span style="color:black"><o:p></o:p></span></font></p>
</div>
</div><p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt;color:black"> <o:p></o:p></span></font></p>
<div><p class="MsoNormal"><span class="apple-style-span"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt;color:black">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></font></span><font color="black"><span style="color:black"><o:p></o:p></span></font></p>
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_______________________________________________ cisco-voip mailing list<span class="apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><span class="apple-converted-space"> </span><a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></span></font></p>
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