I am having a problem with DTMF in a Conference Bride.
This is the call flow.
PSTN FXO/PRI -> PSTN Router -> SIP Dialpeer -> CUCM -> SCCP Phone -> (Conference) -> (IVR) VXML Gateway.
The conference Bridge is the PSTN Router PVDM. this is the RTP audio flow.
PSTN CFB <-----> VXML (IOS) Gateway
When the 3 people (caller, called and IVR) are in conference, PSTN caller press digits but IVR does not recognize it, but we are able to hear the DTFM tones.
We use rtp-nte in out SIP Dial-peer --> CUCM.
How can the CFB Device send DTMF to VXML IOS GW? (its audio is point to point). is the CFB suposed to MIX the RTP NTE and send DTMF?
We still don't have an answer from this, just FYI.
I was thinking some SIP class profiles, to manipulate the reinvites SDP capabilities from the Router to the CUCM.
Which is the limitation, it is a hardware limitation or a software limitation.?
Good afternoon Juan,
I am coming across the same issue and I ran into your post while doing some research. Just curious, did you ever find a resolution and/or workaround to this?
If so, any suggestions will be greatly appreciated!
We worked with Cisco TAC and they created a new CUCM ES version specifically for this issue.
We had to upgrade ALL the nodes in ALL the clusters, and the error went away.
What the new version do is that it invokes software MTPs to subtract the rtp-nte dtmf payload before sending it into a conference resource. In that way, all the participants can press digits and CUCM will intercept them before mixing the audio.
This is less ortodox, because if the MTPs are not locally provided, the audio has to travel all the MPLS just to have DTMF support.
But if you ask me, you can try first enabling SIP-KPML dtmf on all your dial-peers and Trunks, and test that.
KPML travels with the signaling data. not the RTP.
Great, thanks for the feedback Juan. I will give that a try and see if that resolves it. If it doesn't work, more then likely we will proceed with opening up a TAC case.