04-22-2013 08:32 AM - edited 03-18-2019 12:58 AM
Hello Team,
We have added a new Telepresence Endpoint (CTS 1100) to our Telepresence Endpoint pool. We have got Verizon Open Video setup on our CUCM 8.5.1 cluster. We have setup SIP Trunks in CUCM, to reach Verizon Open Video Conference. We have been told by Verizon, that to reach Open Video Conference bridge, we can either dial 1-866-845-273325163192 or a SIP URI 25163192@join.verizon.com. I have no idea on SIP URI setup. Please share your thoughts, so that I will go ahead and configure the necessary details in to CUCM for SIP URI dialing, so that users can dial SIP URI through Ipads or through andrioid devices.All Telepresence endpoints were registered to CUCM cluster.
Thanks,
Solomon.
04-22-2013 02:42 PM
URI dialling is not supported in 8.5.1 - you need to upgrade to 9.x - not sure if it was introduced in 9.0 or if it was 9.1 - check with TAC.
/jens
04-22-2013 09:02 PM
If its more a static config (only specifc addresses should be dialed)
you should be able to use a SIP GW which can rewrite addresses (like the VCS using transforms, search rules, enum, ...).
Your users would still dial only a number like 1234 and that would be mapped to 5678@whateverdomain.com.
Please remember to rate helpful responses and identify
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide