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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I guess by default A-law is enabled in cucm 8.5 , you need to enforce u –law for your calls to ride on 711u-law & this will also avoid use of MTP. even this
works for ivr as well<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">With your setup you may not need to use A-law.<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">Thx<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">manju<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>
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<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>Eric Pedersen<br>
<b>Sent:</b> Wednesday, September 26, 2012 8:29 PM<br>
<b>To:</b> abbas Wali; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] MTP resources Codecs mismatch and DTMF relay<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I'm pretty sure you'll always need MTPs with RFC 2833 because DTMF is carried in the RTP stream. Can you use KPML on your SIP trunks to the IVRs?
<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>
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<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> 25 September 2012 11:55 AM<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>
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<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:l1 level1 lfo3">
<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:l1 level1 lfo3">
<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>
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<span style="font-size:13.5pt;font-family:"Comic Sans MS";color:#000099">@bbas.. </span>
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