We've just implemented a small proof of concept with a Cisco VCS, TelePresence Conductor and 2 8710 TelePresence Server blades.
I've configured a meet.* search rule on the VCS that points to conductor. This is all working properly.
The client now asks me if it's possible to let a audio only participant call into this meeting (from a Cisco Deskphone or a mobilephone via the existing ISDN gateway blade.
There is a CUCM trunk active but I don't see how someone can dial a SIP URI by phone. It's an older CUCM version so URI dialing is not implemented.
Is there some way to create an transform to accomplish this?
Thank you in advance!