debug voip ccapi inout and make sure you're matching an incoming and outgoing dial peer correctly. You can search for peer= to find them (It's right under the big block ccapi creates).
You can also remove cisco-rtp, since that's proprietary and nobody uses it.
There have been some interop issues with the way we do RFC 2833 by default, you may want to add "dtmf-interworking" to the dial peer to see if that changes things.
Another common problem is the other end will advertise a payload type for 2833 that we use for something else, like 100 with NSEs. You can look at the 200 OK they are sending back and see if they're sending 101 in the m-line or not. You should see a a=fmtp 101 1-16 or something similar. If that number is 100 for example, you can do a 'show voice dial-peer 100' and it will list the static payloads for all the codec types. Then change it with the 'rtp payload-type '.
And actually the debug Paolo mentioned is broken (for some call flows) and isn't suggested. 'debug voip rtp session named' is preferred.
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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
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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 ...