If an MX-One TSE SIP extension initiates an outgoing call through a SIP trunk will that use any of the RTP resources? is direct media enabled in this mode?
OK seems like there are two RTP resources used just for setting up the call, a quick test with two MX-One TSE nodes connected together through SIP routes, the SIP phones always receiving the call consumes two IP resources, the SIP extension initiating the call does not consume any resources from it's node !!!!
I have the same experience. No RTP resources are used in the originating exchange, 2 resources are used in the terminating exchange. I have made trace on signalling and it looks like the calling phone got only IP address of IPLU in term.exch. and never got IP from answering phone.
Maybe I did something wrong, but H.323 works OK.
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