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DTMF Sequence is passing as RAW Audio tones instead of RTP Events.

chandra_827
Level 1
Level 1

Hi All,

 

I have a issue where the call flow is as below. When i try to establish the call to third party Audio conference through preconfigured user option from MCU i get only raw DTMF tones in the call instead of RTP events. The Sonexis server will either support KPML or Inband RFC2833 with DTMF as RTP events.

 

Call to XXXXXX conference bridge configured on MCU with a predefined endpoint. The predefined end point is set with dialing number and a DTMF sequence.

 

MCU is registering to VCS as H323 enpoint. VCS has a SIP trunk to CUCM. CUCM intern has a SIP trunk to Sonexis Audio Conference server.

 

Calls to MCU bridge auto dial the number XXXX and go throught the VCS Neighbour zone which is CUCM via SIP trunk. CUCM then negotiates the incoming VCS trunk to out going Sonexis SIP trunk with DTMF RFC 2833 method. Call establishes and then DTMF is RAW tones.

 

When we call the number XXXX (the same number which is preconfigured) that gets connected to Sonexis and the DTMF manually pressed reaches the Sonexis as RTP events. Please Help

MCU--H323 Device-->Registered to VCS>>SIP trunk to>>>CUCM>>SIP Trunk to Sonexis Server.

 

3 Replies 3

Anthony Holloway
Cisco Employee
Cisco Employee

I'm sorry, but I don't fully understand what you described.  Can you explain it in a different way, or use visual aids?

I would recommend looking at the SIP messages to see what is being negotiated end to end.  Alos, keep in mind that CUCM cannot convert from KPML to RTP-NTE without invoking an MTP.  And MTPs don't support video.  So, it either has to be KPML all the way through, or RTP-NTE all the way through, if you want DTMF relay support.

Also, what DTMF do you have configured on your SIP trunks in CUCM?  It should be No Preference for the best interop.

Hi Anthony,

 

Thank you for responding. The call that is initiated by MCU is Audio conference. End to End the call is negotiated RFC2833.

the Video Bridge is on MCU. MCU registers to VCS as H323 End point.

Now MCU is setup with a pre configured dialing for Conferencing Video conference with Audio conference for Audio only participants.

 

When MCU Auto dials the Audio Bridge, the call is answered by Audio conferencing server, however the DTMF sequence that MCU sends are received as RAW Tones instead of RTP events.

 

I have done multiple combination of Sip trunk settings none helped me.

Is it possible for you to post the call setup SIP messages from CUCM perspective so we can see what it receives and what it sends?  Feel free to remove sensitive data.

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