<div dir="ltr">This may be splitting hair here, but two things Ryan:<div><br></div><div>1) Your first sentence reads to me like a contradiction. Could you clarify what you're stating here?</div><div><br></div><div>2) Media can flow through, and even terminate on CUCM, since things like Conference Bridges, MTPs and now the new IVR media resource, are all doing that.</div></div><br><div class="gmail_quote"><div dir="ltr">On Fri, Jul 21, 2017 at 2:53 PM Ryan Huff <<a href="mailto:ryanhuff@outlook.com">ryanhuff@outlook.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<div>So the actual media (RTP) will never flow through a CUCM server; it may however, terminate a connected media stream on a software based MTP application that CUCM runs as a service (IP Voice Media Streaming Application). Signaling (SIP) on the other hand,
will always traverse a CUCM server.</div>
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<div id="m_4427419581952671868AppleMailSignature">If you see a CUCM IP address in the Audio field of the SDP, then it's likely terminating on a CUCM based MTP resource (most often, due to some differences in DTMF negotiations or because the egress path in CUCM is required to use
MTP).</div>
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<div id="m_4427419581952671868AppleMailSignature">If you are trying to test a call using a CUCM MTP resource on a particular cluster node; the simplest way would be to create a new MRG/MRGL that only specifies MTP resources from the desired cluster node and then advertise that
MRGL to the phone and/or egress path to the pstn for the phone and then "require" MTP termination from the phone or egress path.<br>
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Is the problem you're troubleshooting have anything to do with one-way or no-way audio by chance?</div>
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Thanks,</div>
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<div id="m_4427419581952671868AppleMailSignature">Ryan</div></div></div><div dir="auto"><div>
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On Jul 21, 2017, at 3:37 PM, ROZA, Ariel <<a href="mailto:Ariel.ROZA@LA.LOGICALIS.COM" target="_blank">Ariel.ROZA@LA.LOGICALIS.COM</a>> wrote:<br>
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Hi, Guys. <br>
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I need to test problems with calls outgoing from an Ip phone to the PSTN through a particular subscriber (as MTP?).
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How can I force them to do that. <br>
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Packet captures show me that, at times, calls go from my phone to the h323 gateway and sometimes they go from my phone to the Sub and then to the gew.<br>
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Obtener <a href="https://aka.ms/ghei36" target="_blank">Outlook para Android</a></div>
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