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.
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). 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. Is the problem you're troubleshooting have anything to do with one-way or no-way audio by chance? Thanks, Ryan On Jul 21, 2017, at 3:37 PM, ROZA, Ariel <[email protected]<mailto:[email protected]>> wrote: Hi, Guys. I need to test problems with calls outgoing from an Ip phone to the PSTN through a particular subscriber (as MTP?). How can I force them to do that. 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. Obtener Outlook para Android<https://aka.ms/ghei36> _______________________________________________ cisco-voip mailing list [email protected]<mailto:[email protected]> https://puck.nether.net/mailman/listinfo/cisco-voip
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