<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Most SIP carriers support T.38 and/or G.711. The reliability of G.711 fax is mostly dependent upon the ITSP's media gateway platform. <div><br></div><div>The company I work for is using Broadworks as the SIP softswitch platform. The translations are built in such a way that calls will route to the appropriate PSAP no matter what SIP trunk they ingress. We look up based on the originating number and determine which PSAP to route to. The only time we have a concern are the customers who want out of LATA numbers. For example, someone physically resides in San Diego but wants L.A. phone numbers natively on their PBX. If they dial 911 the call will route to an L.A. PSAP. Broadworks has ways to accommodate this by assigning a separate number to the SIP trunk when anyone dials an emergency number. In this case, we would assign a native San Diego DN to the SIP trunk and when anyone utilizing the SIP trunk dials 911 the "From" number will always be the local San Diego number and route to the appropriate PSAP. </div><div><div><br></div><div>Some SIP carriers also offer 911 services through Intrado.</div><div><br></div><div><br><div><div>On Nov 24, 2009, at 5:24 AM, STEVEN CASPER wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="WORD-WRAP: break-word; webkit-nbsp-mode: space; webkit-line-break: after-white-space">
<div>We are testing a centralized SIP trunk design with CUBE and so far it seems to work well for basic inbound and outbound calling. We have over 1000 locations that range in size from 4 analog lines to multiple PRIs.Couple of questions about your SIP deployments:</div>
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<div>Are you using SIP Trunk to support Fax? There does not seems to be a good way to ensure G711 is used for both inbound and outbound fax calls.</div>
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<div>How are you handling 911 calls? With a centralized design I think CER would be required.since users have been known to pickup their phones and relocate without letting anyone know.</div>
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<div>Of course both 911 and FAX could be served using separate analog lines or a PRI but that kind of defeats the purpose.</div>
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<div>Steve</div>
<div><br><br>>>> Mark Holloway <<a href="mailto:mh@markholloway.com">mh@markholloway.com</a>> 11/23/2009 7:20 PM >>><br>SIP Trunking works great when designed and deployed correctly. </div>
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<div>On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:</div><br class="Apple-interchange-newline">
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<div style="FONT-SIZE: 10pt; COLOR: rgb(0,0,0); FONT-FAMILY: Verdana">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 style="COLOR: blue; TEXT-DECORATION: underline" href="mailto:chrward@cisco.com">chrward@cisco.com</a>><br>To: "Ted Nugent" <<a style="COLOR: blue; TEXT-DECORATION: underline" href="mailto:tednugent73@gmail.com">tednugent73@gmail.com</a>>, "Cisco VoIPoE List" <<a style="COLOR: blue; TEXT-DECORATION: underline" 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 US/Canada Eastern<br>Subject: Re: [cisco-voip] SIP Trunk Redundancy<br><br>
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<div style="FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: 'Times New Roman', serif"><span style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125); FONT-FAMILY: Calibri, sans-serif">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="FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: 'Times New Roman', serif"><span style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125); FONT-FAMILY: Calibri, sans-serif"></span> </p>
<div style="FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: 'Times New Roman', serif"><span style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125); FONT-FAMILY: Calibri, sans-serif">-Chris</span></div><p class="MsoNormal" style="FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: 'Times New Roman', serif"><span style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125); FONT-FAMILY: Calibri, sans-serif"></span> </p>
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<div style="FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; 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 style="COLOR: blue; TEXT-DECORATION: underline" 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>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="FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: 'Times New Roman', serif"> </p>
<div style="FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; 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>
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