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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'>While I agree with you, if it needs one for transfer or dtmf, it will not allocate one mid-call. <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><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></div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";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> Friday, November 12, 2010 11:29 AM<br><b>To:</b> 'Mark Holloway'<br><b>Cc:</b> cisco-voip@puck.nether.net<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='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Your right, but I don’t need to have the check box to require one on the trunk. It should allocate one from the MRGL on an as needed basis. <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><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"'> Mark Holloway [mailto:mh@markholloway.com] <br><b>Sent:</b> Friday, November 12, 2010 11:28 AM<br><b>To:</b> Bill Riley<br><b>Cc:</b> 'Ryan Ratliff'; cisco-voip@puck.nether.net<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>You should be using an MTP for your SIP trunk to support DTMF. It does not need to be a hardware MTP resource. <o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><div><p class=MsoNormal>On Nov 12, 2010, at 10:05 AM, Bill Riley wrote:<o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>When I change it to MTP required on the sip trunk everything works as expected. The point is that I shouldn’t need to have this configuration.</span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in;border-width:initial;border-color:initial'><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Ryan Ratliff [mailto:rratliff@cisco.com]<span class=apple-converted-space> </span><br><b>Sent:</b><span class=apple-converted-space> </span>Friday, November 12, 2010 9:00 AM<br><b>To:</b><span class=apple-converted-space> </span>Bill Riley<br><b>Cc:</b><span class=apple-converted-space> </span>'Cheng, Karen';<span class=apple-converted-space> </span><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><b>Subject:</b><span class=apple-converted-space> </span>Re: [cisco-voip] SIP trunk one way audio</span><o:p></o:p></p></div></div></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>I see lots of discussion around config and call flow but have you actually done any troubleshooting? One way audio most of the time comes down to the simple fact that one party is not receiving RTP from the other. For these one-way audio calls you need to determine what IP addresses are involved. Next verify this in the signaling via the SDPs in the SIP messages. You can then use show commands on the router to confirm where it thinks it should be sending and receiving RTP to/from and if in fact packet counters are incrementing.<o:p></o:p></p></div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div><div><div><p class=MsoNormal>If no MTP is being used for the call, try forcing it to use one and see if that fixes the issue.<o:p></o:p></p></div></div><div><div><p class=MsoNormal>If you are using media flow-through (default) does changing it to flow-around fix the issue? <o:p></o:p></p></div></div><div><div><p class=MsoNormal> <o:p></o:p></p></div><div><div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif";color:black'>-Ryan</span><o:p></o:p></p></div></div></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><div><div><p class=MsoNormal>On Nov 12, 2010, at 8:09 AM, Bill Riley wrote:<o:p></o:p></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br><br><o:p></o:p></p></div><div><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I shouldn’t need an MTP for this connection. All SIP traffic is sourced from one interface. I do have SCCP traffic sourced from a different interface but it is only used for a conference bridge, not MTP.</span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div></div><div><div style='border:none;border-top:solid windowtext 3.0pt;padding:3.0pt 0in 0in 0in;border-width:initial;border-color:initial;border-width:initial;border-color:initial'><div><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Cheng, Karen [mailto:Karen.Cheng@racq.com.au]<span class=apple-converted-space> </span><br><b>Sent:</b><span class=apple-converted-space> </span>Thursday, November 11, 2010 9:14 PM<br><b>To:</b><span class=apple-converted-space> </span>'Bill Riley'<br><b>Subject:</b><span class=apple-converted-space> </span>RE: [cisco-voip] SIP trunk one way audio</span><o:p></o:p></p></div></div></div></div><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> </span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Not sure if you have checked already but is your SIP trunk using one interface and your MTP/SCCP interface a different interface?</span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I had one-way audio/no audio problems ages back due to this because our integrator had configured the SIP trunk to point to int gi0/0’s IP and then configured the SCCP interface to loopback0.</span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Regards</span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div></div><div><div><div><div><p class=MsoNormal><b><i><span lang=EN-AU style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#003478'>Karen Cheng</span></i></b><o:p></o:p></p></div></div></div><div><div><div><p class=MsoNormal><i><span lang=EN-AU style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#003478'>Voice Network Engineer</span></i><o:p></o:p></p></div></div></div><div><div><div><p class=MsoNormal><span lang=EN-AU style='color:#1F497D'> </span><o:p></o:p></p></div></div></div><div><div><div><div><p class=MsoNormal><span lang=EN-AU style='color:#1F497D'> </span><o:p></o:p></p></div></div></div></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='color:#1F497D'> </span><o:p></o:p></p></div></div><div><div style='border:none;border-top:solid windowtext 3.0pt;padding:3.0pt 0in 0in 0in;border-width:initial;border-color:initial;border-width:initial;border-color:initial'><div><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><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>On Behalf Of<span class=apple-converted-space> </span></b>Bill Riley<br><b>Sent:</b><span class=apple-converted-space> </span>Friday, 12 November 2010 2:15 AM<br><b>To:</b><span class=apple-converted-space> </span><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><b>Subject:</b><span class=apple-converted-space> </span>[cisco-voip] SIP trunk one way audio</span><o:p></o:p></p></div></div></div></div><div><div><p class=MsoNormal><span lang=EN-AU style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> </span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>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.</span><o:p></o:p></p></div></div><div><div><p class=MsoNormal><span lang=EN-AU><br></span><span lang=EN-AU style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:green'>75% of inspected vehicle have a defect*. Call 13 1905 and book an RACQ Vehicle Inspection today. *Based on the RACQ Vehicle Defect Report January 2008<br><br></span><span lang=EN-AU><br></span><span lang=EN-AU style='font-size:10.0pt;font-family:"Arial","sans-serif";color:gray'>Please Note: If you are not the intended recipient, please delete this email as its use is prohibited. RACQ does not warrant or represent that this email is free from viruses or defects. If you do not wish to receive any further commercial electronic messages from RACQ please e-mail<span class=apple-converted-space> </span><a href="mailto:unsubscribe@racq.com.au">unsubscribe@racq.com.au</a><span class=apple-converted-space> </span>or contact RACQ on 13 19 05.</span><o:p></o:p></p></div></div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:"Lucida Grande"'>_______________________________________________<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">https://puck.nether.net/mailman/listinfo/cisco-voip</a></span><o:p></o:p></p></div></div></div><div><p class=MsoNormal> <o:p></o:p></p></div></div><p class=MsoNormal><span style='font-size:13.5pt;font-family:"Verdana","sans-serif"'>_______________________________________________<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">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>