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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>One note on the T38. <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'>I’ve hit this with multiple SIP providers where T38 relay is simply not supported. The workaround is to basically send fax calls through a back-to-back T1 card. Take the inbound VoIP call, send it outbound to one side of a T1, it’ll come out the other side, and voila! If you know ahead of time, and have an idea of where the fax server is, this is a pretty easy workaround.<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'>Thanks,<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><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'>James Buchanan </span></b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7E7E7E'>|</span><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'> Technology Manager, UC </span></b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7E7E7E'>|</span><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'> <b>South Region </b></span><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7E7E7E'>|</span><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'> Presidio Networked Solutions <br>12 Cadillac Dr, Suite 130, Brentwood, TN 37027 </span></b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7E7E7E'>|</span><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'> </span></b><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7E7E7E'><a href="mailto:jbuchanan@ctiusa.com">jbuchanan@presidio.com</a><br></span></b><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'>D: 615-866-5729</span></b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'> </span><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7E7E7E'>| <b>F:</b> </span><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'>615-866-5781</span></b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'> | </span><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7E7E7E'><a href="http://www.presidio.com/">www.presidio.com</a></span></b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> <o:p></o:p></span></p><p class=MsoNormal><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'>CCIE #25863, Voice</span></b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><br><br></span><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'><o:p> </o:p></span></p><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"'> cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Tony Edwards<br><b>Sent:</b> Monday, February 14, 2011 3:02 AM<br><b>To:</b> Lisa Notarianni<br><b>Cc:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> Re: [cisco-voip] Verizon SIP vs. PRI?<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>here is my limited understanding..<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>1) sip / cube trunk is certainly looks promising , however , when you build a business case , make sure you add up any wan bw upgrade costs to accommodate x number of concurrent g711 calls for example. with all overhead , some telcos recommend 100k for a g711 call , so you got to budget 500k for rtp streams straight away on the same wan pipe , where your data apps will flow. so , the costs savings of not buying e1 or t1 trunk & not buying e1 or t1 controller cards on the voice gw can easily disappear , if you do maths for sip trunk wan bw upgrade.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>2) secondly , I gather they still have some having issues , even though theoretically , t.38 should work well on this trunk. so you got to talk with telco closely with their own offerings. in a worst case scenario , you can still run some vg based faxing or even direct pstn if you few fax machines on a given site.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>3) watch out issues with sip profiles , which are essentially like voice translation rules on h323 gw, where you match & replace digits. depending upon telco , you got to send your ani with the realm name , rather than cube's ip address.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>4) depending upon flow around or flow through deployment , you need to configure mtp at ccm.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>5) also , some times i have seen issues where sip trunk in ccm pointing to the loop int of cube. it got fixed when i changed to its wan interface at one of my cube cut overs.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>finally , my biggest concern is the number portability of pstn & isdn in to sip platform , which i gather is still an issue with some of the telcos around the world.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>also do not forget to configure the wan qos on 5060 port for sip traffic when you role out a cube on production network.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>other than that , i reckon sip cube trunks are very valid alternative to $$ expensive pri trunks.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>tony<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal><br><br> <o:p></o:p></p></div><div><p class=MsoNormal>On Wed, Feb 9, 2011 at 6:13 AM, Lisa Notarianni <<a href="mailto:notariannil1@scranton.edu">notariannil1@scranton.edu</a>> wrote:<o:p></o:p></p><div><p class=MsoNormal>We currently utilize PRI trunks as fail over and backup. They connect to our 6509's.<br><br>Can anyone share their thoughts on SIP vs. PRI services. <br><br>Thanks,<br><br>Lisa<o:p></o:p></p><div><p class=MsoNormal>-- <br><img border=0 width=250 height=100 id="_x0000_i1025" src="cid:image001.jpg@01CBCC20.99F2D840"><o:p></o:p></p></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>