CME 8.0 (15.0(1)XA) vs SIP trunk DTMF issues

Unanswered Question
Nov 28th, 2009
User Badges:


I've upgraded my 2821 to 15.0(1)XA to try out CME 8.0. I have a SIP trunk coming in from an ITSP and everything works fine using 12.4(22)YB4. When I upgrade to 15.0(1)XA, DTMF stops working on 90% of my calls (I am forcing RTP-NTE). I have not changed the config between 12.4(22)YB4 and 15.0(1)XA at all.

Tech support tells me that they are receiving RFC2833 packets but they're too short - according to them, the 3 first packets have 0ms duration, the middle packet has 400ms duration and the last 3 have 800ms. They have suggested that I send inband DTMF instead, but I am not really inclined to do that because it worked before. I've tried specifying the payload setting (101) and other settings on my dial-peers to no avail. I have been forced to return to CME 7.1 because of this.

All my phones are SCCP and this problem is reproduced on a broad range of phones including analog phones off a EVM board (SCCP-controlled).

Any ideas?

  • 1
  • 2
  • 3
  • 4
  • 5
Overall Rating: 0 (0 ratings)
Nicholas Matthews Sat, 11/28/2009 - 13:34
User Badges:
  • Red, 2250 points or more

You may want to try this command:

voice service voip

dtmf-interworking rtp-nte

It changes the interarrival properties of the 2833 packets, and we see it particularly with Verizon trunks, though it has been useful in other scenarios as well.


J. S. Black Sat, 12/12/2009 - 19:50
User Badges:

Hi Nick,

I just tested dtmf-interworking with 15.0(1)XA, no change.

Any debugs I should be looking at?

paolo bevilacqua Sat, 11/28/2009 - 23:47
User Badges:
  • Super Gold, 25000 points or more
  • Hall of Fame,

    Founding Member

The rigth approach would be to have cisco fix what is broken, unfortunately the burden of proof is on you and it can take a lot of time and effort.

The alternative is to go back to working IOS and upgrade when new software is less buggy.

Nicholas Matthews Mon, 12/14/2009 - 06:33
User Badges:
  • Red, 2250 points or more

I would start by getting a packet capture.  Then, you can look if you receive the packets.  Something like 'rtp.payload_type == 101' may be a good wireshark filter.

For debugs, you're looking at using 'debug ccsip messages' and 'debug voip rtp session named'.

You're probably best off opening a tac case with this information because it sounds like something that isn't as straightforward.  Double check your dial peers - maybe they were read in differently with the new IOS and haven't been applied the same in the two versions.



This Discussion