We have a new IVR built using call studio and we are using ucce 8.0 and cvp 8.0
Brief setup is as follows:
Location1 CUBE-->SIP Gateway-->MPLS-->Location2 CUBE-->Call server-->ICM-->Call server-->VXML Gateway-->stream ivr rtp from vxml gateway to caller.
When we dial from location1, call is answered, we wont hear any audio but if we press any dtmf input it is routing properly to respective call taking agents and also we can call duration ticking on phone display.
From location2 CUBE if we are translating and routing to call manager and answering on a phone, we are able to hear 2 way audio but it is not the case from ivr is played from vxml gateway.
And the same call studio appliaction works perfectly fine when we are dialling from location3 through another DID pointed to same script.
You have reached the Cisco Logistics Support Center.. To Check Status of
your RMA, visit Product Returns & Replacements (RMA). Need help? Contact
us by Phone or Email. North Americas Phone: 1800 553 2447 Option 4
Email: firstname.lastname@example.org Europe Phone: +3...
The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
the softkeys. You enable or disable them via TCL. Here is the long
answer. Be sure to read the whole thing or e...
Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...