09-26-2014 03:07 AM - edited 03-18-2019 03:27 AM
Hi
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!
Solved! Go to Solution.
09-26-2014 08:11 PM
No, you need a CUCM version which enables URI dialing and endpoints which can do URI dialing.
09-26-2014 08:11 PM
No, you need a CUCM version which enables URI dialing and endpoints which can do URI dialing.
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