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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:10.0pt;font-family:"Arial","sans-serif";color:black'>You mention enabled MTP on those SIP trunks, you can enable it at the trunk or gateway and it will force every call to use MTP. However, if you set it, I believe in the trunk profile, or security profile, cannot recall exactly, but you can set it to only engage the mtp resource if necessary.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'>Dennis Heim</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'>Sr. UC Engineer<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'>World Wide Technology<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'>Office: 314.212.1814<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'>Email: <a href="mailto:dennis.heim@wwt.com"><span style='color:blue'>dennis.heim@wwt.com</span></a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'><a href="http://www.wwt.com/"><span style='color:blue'>www.wwt.com</span></a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'><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>abbas Wali<br><b>Sent:</b> Tuesday, September 25, 2012 1:55 PM<br><b>To:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> [cisco-voip] MTP resources Codecs mismatch and DTMF relay<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Recently we had a major VOIP fault which was escalated to TAC. Whom also spent some good time before realizing that we were running out of MTP resources at busy times. <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><o:p> </o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>What was happening that calls coming on phones and as soon as the receiver was lifted, the called party was hearing a fast busy. <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Initially we had to restart the whole cluster and it used to settled down. <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>In the end we have to increase the MTP resources in the Service Parameters from 24 to 48 for each of the nodes, since then this fault has gone away. <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Just going back into recent past –we believed that this issue started after we enabled MTP resource requirements for the SIP trunks, who at that time were struggling with the DTMF relays. So by enabling it for the SIP trunks we got into contentions for the rest of the network. <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>My question is (and I have overheard somewhere) that if your cluster is configured properly, you will never require the use of MTP resources (or may be in a very lesser amount). Also I feel like some mis configuration on the SIP trunks which has forced us to get help from the MTP resources to support the DTMF relays. <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Below are some of the points which emphasizes my points<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><ol start=1 type=1><li class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1'><span style='font-size:7.0pt'> </span>PCM type for our E1 cards are set to A-Law but when we make an VOIP to VOIP call internally it shows the codec as ULaw. (so to me it looks like that we are using MTP to convert from A to U for incoming PSTN calls)<o:p></o:p></li><li class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1'><b><span style='font-size:7.0pt'> </span></b>For the DTMF –the gateways are configured with <b><u>mgcp dtmf-relay voip codec all mode out-of-band </u></b>while some of the SIP trunks are set to RFC 2833. I believe these are two different modes which again requires MTP for translation. <o:p></o:p></li></ol><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b> </b><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I believe if we make the codecs even by forcing the CUCM to use A Law internally as well and also use one single mode for the DTMF relays then we should not require to increase the MTP resources.<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Just to add we are using CUCM 8.5. we have some 7k phones. we have MGCP gateways. And SIP trunks to 3<sup>rd</sup> party IVR’s and another neighbour CUCM 7.<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Any help will be appreciated. <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><br clear=all><br>-- <br><span style='font-size:13.5pt;font-family:"Comic Sans MS";color:#000099'>@bbas.. </span><o:p></o:p></p></div></body></html>