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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='color:#1F497D'>Well according to TAC if the call is sip to sip I shouldn’t need the MTP at all.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>I thought MTP required check box allocated the MTP if it needed one or not and leaving it unchecked allocated the MTP if it needed to use one.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><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"'> Buchanan, James [mailto:jbuchanan@presidio.com] <br><b>Sent:</b> Thursday, November 11, 2010 11:49 AM<br><b>To:</b> Bill Riley<br><b>Subject:</b> RE: [cisco-voip] SIP trunk one way audio<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='color:#1F497D'>Because it won’t invoke the MTP without it.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><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 style='margin-bottom:12.0pt'><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'>CCIE #25863, Voice</span></b><span style='color:#1F497D'><o:p></o:p></span></p></div><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><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"'> Bill Riley [mailto:bill@hitechconnection.net] <br><b>Sent:</b> Thursday, November 11, 2010 11:19 AM<br><b>To:</b> Buchanan, James<br><b>Subject:</b> RE: [cisco-voip] SIP trunk one way audio<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='color:#1F497D'>No I don’t have “MTP Required” Checked. The MTP’s are set to only use G.711. I don’t see why I need to have MTP Required on this trunk. <o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><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"'> Buchanan, James [mailto:jbuchanan@presidio.com] <br><b>Sent:</b> Thursday, November 11, 2010 11:12 AM<br><b>To:</b> Bill Riley<br><b>Subject:</b> RE: [cisco-voip] SIP trunk one way audio<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='color:#1F497D'>Dumb question—I assume you have “MTP Required” checked, are set to use only G711 to the MTP, and reset the SIP trunk after setting it? If you require G729, you will need to setup an IOS MTP. These do not use DSPs, but will allow other codecs besides G711. <o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><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 style='margin-bottom:12.0pt'><b><span style='font-size:7.0pt;font-family:"Verdana","sans-serif";color:#7F7F7F'>CCIE #25863, Voice</span></b><span style='color:#1F497D'><o:p></o:p></span></p></div><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><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"'> cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Bill Riley<br><b>Sent:</b> Thursday, November 11, 2010 10:15 AM<br><b>To:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> [cisco-voip] SIP trunk one way audio<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I have a new SIP trunk terminating in a 2921 CUBE bundle. When I call in from the Trunk directly to an IP phone it works correctly. If I call from the trunk to IP phone and the IP phone transfers the call without waiting for the remote party to answer I get one way audio. From reading this looks like and MTP issue but I have an MTP set in the MRGL for the trunk. <o:p></o:p></p></div></body></html>