Calls input problem through SIP Gateway CUCM 11.5 VoIP Phone 7821

[Paki]

Solution is requested for following problem.

Connectivity Scenario:

VoIP Phone (c7821) --> CUCM (11.5) --> SIP Gateway Router (3945) --> ITSP

Problem Statement:

We are unable to send call input after a call is connected to a remote IVR (outbound calls); i.e. UAN Numbers or Service Numbers. 

Self Troubleshooting:

To rectify the problem, I have tried dtmf-relay rtp-nte, sip-notify, sip-info & sip-kpml alongwith MTP checked & unchecked & DTMP Preference RFC2833 & RFC2833 and OOB turn by turn.

Though for RFC2833, OOB & In-band are auto-adjustable over SIP Trunk & Gateway but I am totally unable to troubleshoot the problem.

Help from professionals is requested please.

Thank you

[TONY SMITH]

First question is what protocol and options does the ITSP use for DTMF relay.  Normally it's rtp-nte on a SIP trunk, or at least that's what I've seen.   However some use different variants, for example we have one install where the ITSP uses type 97 rather than 101.  This can be seen in their SDP as "a=fmtp:97 0-15".   Different types can be fixed up in the dial peers, 

[Paki]

ITSP is using rtp-nte (option 101) as DTMF - RELAY PAYLOAD.

Also, please find attached herewith traces for inbound call from my cell phone to my office number.

Thank you for your valuable time.

[TONY SMITH]

I think you want to look at SIP debugs "debug ccsip mess" to have a look at the signalling both between CUCM and the gateway and between the gateway and ITSP.   Specific debug to look at the actual rtp-nte is "debug voip rtp session named"