If I wanted the CM to be used as Evolution Server (with H.323 phones registering it to), I'm guessing I would have to configure it as a SIP Entity of "CM" and then it would be applicable to configure the CM for origination and terminate sequences for the user profile. I thought this would be a simpler setup and later I would migrate to a CM as Evolution Server configuration. (None) is selected for both origination and termination application sequence for the user profile, the reason being is that I configured the CM as a SIP entity of "SIP trunk" because for the initial phase, I wanted it to be treated as such. I can supply any relevant information as requested.
How can I make the SM not care about the originating user of an inbound call from a SIP trunk to the SM?Īny advice would be greatly appreciated. Note, at this point CM should just be viewed as the other end of a SIP trunk and not an evolution or feature server. I've tried to pare down the routing configuration on the SM to what's minimally necessary to make the work. I haven't been able to figure out why the SM returns the "404 Not Found (originating user not found)". However, the SM ultimately responds with "404 Not Found (originating user not found)" to CM, despite the SIP extension being called to is SIP registered to the SM when the inbound call is placed.įurthermore, if I perform a Call Routing test from SMGR using the Called Party URI, Calling Party URI, Calling Party Address for the failed call obtained extracted from traceSM on the SM and use the appropriate Called Session Manager Instance, Session Manager Listen Port, and Transport Protocol, the result of the executed Call Routing Test indicates the call would be forwarded to the enduser, i.e., "Proxying the request to endpoint 89584." which is what I want to happen for the failed call. When I make an inbound call to this extension, a SIP Invite is send from the CM to the SM (verified from traceSM on SM). One extension is configured on the CM to be be routed across the SIP trunk to the SM. I've setup a SIP trunk to our CM and wanted to test inbound calls from the CM to the SM. I've been able to provision and register SIP endpoints to the SM and make calls between these SIP endpoints. I installed Session Manager (SM 6.3 SP 8) for testing before it is deployed in production to replace our SES.
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