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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'>I am not saying you don’t need an MTP. I am saying you don’t need to use the MTP required check box. If an MTP is required it should allocate one from the MRGL. <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> Monday, November 15, 2010 9:59 AM<br><b>To:</b> Jason Aarons (US)<br><b>Cc:</b> Bill Riley; 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>Assuming you aren't using a SIP phone load, then when generating a DTMF tone from the SIP trunk towards a SIP provider and using an out of band method, the trunk needs the resource to generate the out of band tone. In CUCM 7 if you don't allocated the MTP to the SIP trunk DTMF does not work. I have not tried to do this by strictly relying on CUBE, which may supplement it, but it may also depend if you are using SDP transparency.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Nov 15, 2010, at 8:49 AM, Jason Aarons (US) wrote:<o:p></o:p></p></div><p class=MsoNormal><br><br><o:p></o:p></p><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>If using RFC2833 (rtp-nte) which both 7900 SIP LOAD and IOS and Unity support, why is MTP Required checkmark required technically? What’s happening that needs MTP?</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"'><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>Mark Holloway<br><b>Sent:</b><span class=apple-converted-space> </span>Monday, November 15, 2010 10:34 AM<br><b>To:</b><span class=apple-converted-space> </span>Bill Riley<br><b>Cc:</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>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>Unless you are using inband DTMF it will be required.<o:p></o:p></p></div><div><div><p class=MsoNormal> <o:p></o:p></p></div><div><div><div><p class=MsoNormal>On Nov 15, 2010, at 7:07 AM, Bill Riley wrote:<o:p></o:p></p></div></div><div><p class=MsoNormal><br><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 don’t think that’s correct. It needs access to an MTP but you shouldn’t need to use the MTP required checkbox on the trunk.</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><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"'>Mark Holloway [mailto:mh@markholloway.com]<span class=apple-converted-space> </span><br><b>Sent:</b><span class=apple-converted-space> </span>Friday, November 12, 2010 12:24 PM<br><b>To:</b><span class=apple-converted-space> </span>Bill Riley<br><b>Cc:</b><span class=apple-converted-space> </span>'Ryan Ratliff';<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><div><div><p class=MsoNormal> <o:p></o:p></p></div></div><div><div><p class=MsoNormal>CUCM needs it assigned to the trunk for DTMF to work properly for calls egressing the SIP Trunk.<o:p></o:p></p></div></div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div><div><div><div><div><div><p class=MsoNormal>On Nov 12, 2010, at 10:50 AM, Bill Riley wrote:<o:p></o:p></p></div></div></div><div><div><p class=MsoNormal><br><br><br><br><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";color:#1F497D'>I don’t think that is correct. It should only need to have one available in the MRGL, not one allocated every time a call comes in.</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";color:#1F497D'> </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";color:#1F497D'> </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";color:#1F497D'>voice service voip</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";color:#1F497D'>ip address trusted list</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";color:#1F497D'> ipv4 x.x.x.x</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";color:#1F497D'> ipv4 x.x.x.x</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";color:#1F497D'> ipv4 x.x.x.x</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";color:#1F497D'>allow-connections h323 to h323</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";color:#1F497D'>allow-connections h323 to sip</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";color:#1F497D'>allow-connections sip to h323</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";color:#1F497D'>allow-connections sip to sip</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";color:#1F497D'>no supplementary-service sip moved-temporarily</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";color:#1F497D'>no supplementary-service sip refer</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";color:#1F497D'>fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback none</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";color:#1F497D'>h323</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";color:#1F497D'>modem passthrough nse codec g711ulaw</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";color:#1F497D'>sip</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";color:#1F497D'> bind control source-interface Serial0/0/0:1</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";color:#1F497D'> bind media source-interface Serial0/0/0:1</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";color:#1F497D'> early-offer forced</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";color:#1F497D'> midcall-signaling passthru</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";color:#1F497D'>!</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";color:#1F497D'> </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";color:#1F497D'> </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";color:#1F497D'>voice class codec 100</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";color:#1F497D'>codec preference 1 g711ulaw</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";color:#1F497D'>codec preference 2 g729r8</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";color:#1F497D'>!</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";color:#1F497D'>! </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";color:#1F497D'>dial-peer voice 201 voip</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";color:#1F497D'>preference 1</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";color:#1F497D'>destination-pattern 91[2-9]..[2-9]......</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";color:#1F497D'>rtp payload-type cisco-codec-fax-ind 98</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";color:#1F497D'>rtp payload-type comfort-noise 13</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";color:#1F497D'>session protocol sipv2</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";color:#1F497D'>session target ipv4: x.x.x.x</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";color:#1F497D'>dtmf-relay rtp-nte</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";color:#1F497D'>codec g711ulaw</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";color:#1F497D'>fax rate 14400</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";color:#1F497D'>ip qos dscp cs3 signaling</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";color:#1F497D'>!</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";color:#1F497D'>dial-peer voice 202 voip</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";color:#1F497D'>preference 1</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";color:#1F497D'>destination-pattern 9[2-9]......T</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";color:#1F497D'>rtp payload-type cisco-codec-fax-ind 98</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";color:#1F497D'>rtp payload-type comfort-noise 13</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";color:#1F497D'>session protocol sipv2</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";color:#1F497D'>session target ipv4: x.x.x.x</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";color:#1F497D'>dtmf-relay rtp-nte</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";color:#1F497D'>codec g711ulaw</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";color:#1F497D'>fax rate 14400</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";color:#1F497D'>ip qos dscp ef signaling</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";color:#1F497D'>!</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";color:#1F497D'>dial-peer voice 203 voip</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";color:#1F497D'>preference 1</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";color:#1F497D'>destination-pattern ^9556[2-9]......</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";color:#1F497D'>rtp payload-type cisco-codec-fax-ind 98</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";color:#1F497D'>rtp payload-type comfort-noise 13</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";color:#1F497D'>session protocol sipv2</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";color:#1F497D'>session target ipv4: x.x.x.x</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";color:#1F497D'>dtmf-relay rtp-nte</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";color:#1F497D'>codec g711ulaw</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";color:#1F497D'>fax rate 14400</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";color:#1F497D'>ip qos dscp ef signaling</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";color:#1F497D'>!</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";color:#1F497D'>dial-peer voice 204 voip</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";color:#1F497D'>preference 1</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";color:#1F497D'>destination-pattern 9555[2-9]......</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";color:#1F497D'>rtp payload-type cisco-codec-fax-ind 98</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";color:#1F497D'>rtp payload-type comfort-noise 13</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";color:#1F497D'>session protocol sipv2</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";color:#1F497D'>session target ipv4: x.x.x.x</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";color:#1F497D'>dtmf-relay rtp-nte</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";color:#1F497D'>fax rate 14400</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";color:#1F497D'>ip qos dscp ef signaling</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";color:#1F497D'>!</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";color:#1F497D'>dial-peer voice 411 voip</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";color:#1F497D'>preference 1</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";color:#1F497D'>destination-pattern 5555555..</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";color:#1F497D'>rtp payload-type cisco-codec-fax-ind 98</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";color:#1F497D'>rtp payload-type comfort-noise 13</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";color:#1F497D'>session protocol sipv2</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";color:#1F497D'>session target ipv4: x.x.x.x</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";color:#1F497D'>dtmf-relay rtp-nte</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";color:#1F497D'>codec g711ulaw</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";color:#1F497D'>fax rate 14400</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";color:#1F497D'>ip qos dscp cs3 signaling</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";color:#1F497D'>! </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";color:#1F497D'>dial-peer voice 412 voip</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";color:#1F497D'>preference 2</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";color:#1F497D'>destination-pattern 5555555..</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";color:#1F497D'>rtp payload-type cisco-codec-fax-ind 98</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";color:#1F497D'>rtp payload-type comfort-noise 13</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";color:#1F497D'>session protocol sipv2</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";color:#1F497D'>session target ipv4: x.x.x.x</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";color:#1F497D'>dtmf-relay rtp-nte</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";color:#1F497D'>codec g711ulaw</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";color:#1F497D'>fax rate 14400</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";color:#1F497D'>ip qos dscp cs3 signaling</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";color:#1F497D'>!</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";color:#1F497D'>dial-peer voice 413 voip</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";color:#1F497D'>preference 3</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";color:#1F497D'>destination-pattern 5555555..</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";color:#1F497D'>rtp payload-type cisco-codec-fax-ind 98</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";color:#1F497D'>rtp payload-type comfort-noise 13</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";color:#1F497D'>session protocol sipv2</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";color:#1F497D'>session target ipv4: x.x.x.x</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";color:#1F497D'>dtmf-relay rtp-nte</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";color:#1F497D'>codec g711ulaw</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";color:#1F497D'>fax rate 14400</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";color:#1F497D'>ip qos dscp cs3 signaling</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";color:#1F497D'> </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";color:#1F497D'> </span><o:p></o:p></p></div></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;border-width:initial;border-color:initial'><div><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"'>Mark Holloway [mailto:mh@markholloway.com]<span class=apple-converted-space> </span><br><b>Sent:</b><span class=apple-converted-space> </span>Friday, November 12, 2010 11:32 AM<br><b>To:</b><span class=apple-converted-space> </span>Bill Riley<br><b>Cc:</b><span class=apple-converted-space> </span>'Ryan Ratliff';<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></div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div><div><div><div><p class=MsoNormal>You should have it on the trunk.<o:p></o:p></p></div></div></div><div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div><div><div><div><div><div><p class=MsoNormal>On Nov 12, 2010, at 10:29 AM, Bill Riley wrote:<o:p></o:p></p></div></div></div></div><div><div><div><p class=MsoNormal><br><br><br><br><br><o:p></o:p></p></div></div></div><div><div><div><div><div><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.</span><o:p></o:p></p></div></div></div></div><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><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;border-width:initial;border-color:initial;border-width:initial;border-color:initial'><div><div><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"'>Mark Holloway [mailto:mh@markholloway.com]<span class=apple-converted-space> </span><br><b>Sent:</b><span class=apple-converted-space> </span>Friday, November 12, 2010 11:28 AM<br><b>To:</b><span class=apple-converted-space> </span>Bill Riley<br><b>Cc:</b><span class=apple-converted-space> </span>'Ryan Ratliff';<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></div></div><div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div></div><div><div><div><div><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></div></div></div><div><div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div></div></div><div><div><div><div><div><div><div><p class=MsoNormal>On Nov 12, 2010, at 10:05 AM, Bill Riley wrote:<o:p></o:p></p></div></div></div></div></div><div><div><div><div><p class=MsoNormal><br><br><br><br><br><br><o:p></o:p></p></div></div></div></div><div><div><div><div><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></div></div></div><div><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></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;border-width:initial;border-color:initial;border-width:initial;border-color:initial;border-width:initial;border-color:initial'><div><div><div><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"'>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></div></div></div><div><div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div></div></div><div><div><div><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></div></div><div><div><div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div></div></div></div><div><div><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></div></div><div><div><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></div></div><div><div><div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div></div></div><div><div><div><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></div></div></div><div><div><div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div></div></div></div><div><div><div><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></div></div></div><div><div><div><div><div><p class=MsoNormal><br><br><br><br><br><br><br><o:p></o:p></p></div></div></div></div></div><div><div><div><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></div></div><div><div><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></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;border-width:initial;border-color:initial;border-width:initial;border-color:initial;border-width:initial;border-color:initial;border-width:initial;border-color:initial'><div><div><div><div><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></div></div><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></div></div><div><div><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></div></div><div><div><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><div><div><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></div></div><div><div><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><div><div><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></div></div><div><div><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><div><div><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></div><div><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></div><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><div><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></div></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><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;border-width:initial;border-color:initial;border-width:initial;border-color:initial;border-width:initial;border-color:initial;border-width:initial;border-color:initial'><div><div><div><div><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></div></div><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></div></div><div><div><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></div></div><div><div><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. 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