DTMF issue on CME

[raid_t]

Hi all,

We have a CME (C2851) with IOS 12.4(13r)T11, RELEASE SOFTWARE (fc1). A SIP Trunk is configured with ITSP. Calls work fine, inbound and outbound, however  typing the digits is not working eventhough we hear the tone when typed.

IP Phones are C7911.

Collected debugs and could see that ITSP uses raw inband.

SCCP phones does out-of-band DTMF?

Out of Band to Inband DTMF conversion is not supported?

Tried adding "dtmf-interworking rtp-nte" command, changing to different "dtmf-relay" options under dial-peer section.

Do you think it is possible to resolve this on our side? Or we check with ITSP if they could negotiated RTP-NTE or out of band DTMF for the call?

Thanks!

R.

[raid_t]

Can anyone shed some light here please?

Thanks!

R.

[Chris Deren]

Post your config please.

[raid_t]

Hi Chris,

Attached the relevant config. If you need any additional section, please let me know.

Thanks.

R.

[raid_t]

bump.

Anyone?

[Chris Deren]

Change your dial peer with the following:

dial-peer voice 42 voip

incoming called-numer .

dtmf-relay rtp-nte sip-kpml

[raid_t]

Hi Chris,

Dial peer 42 is the outbound and tried your recommendation on this dial peer but it didn't work.

Incoming dial peer is as below:

dial-peer voice 40 voip
 corlist outgoing CORLIST-EXTERNAL
 description ***Incoming Dial-Peer***
 translation-profile incoming SIP_IN
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 incoming called-number +xxxxxxxxxxx
 dtmf-relay rtp-nte
 no vad
!

regards,

R.

[Chris Deren]

OK, can you add the following to dial-peer 40 then:

dtmf-relay rtp-nte sip-kpml

[raid_t]

I can try that, but as far as I see this has to do with incoming calls right? The issue we're facing is for outgoing calls, that's why I showed the configuration for outgoing dial peer.

From what I read on other similar cases the SCCP doesn't support in band DTMF thus a conversion is needed with DSP. DSP was already in place.

Sorry if I wasn't clear enough on describing the issue.

Thanks,

R.

[Chris Deren]

your incoming dial-peer is matched on outbound calls as it is being used when call hits the GW from CUCM side. Remember every call matches 2 dial-peers inbound and outbound.

[raid_t]

just to be on same page I'd also like to let you know the call flow:

Call flow: SCCP Phone c7911 à CME (C2851) à SIP à ITSP

I'll make the recommended change to inbound dial peer and let you know the result.

Thanks,

R.

[raid_t]

Hi Chris,
I just added the command to the incoming dial peer but it's still not working! Can only hear the beeps when digits are pressed.

edit: posting also this output below from debug ccsip messages if that helps:

INVITE:

v=0
o=CiscoSystemsSIP-GW-UserAgent 3000 5718 IN IP4 x.x.x.x
s=SIP Call
c=IN IP4 x.x.x.x
t=0 0
m=audio 18798 RTP/AVP 8 0 101
c=IN IP4 x.x.x.x
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

UPDATE:

v=0
o=LucentPCSF 685664112 685664113 IN IP4 x.x.x.x
s=-
c=IN IP4 x.x.x.x
t=0 0
m=audio 22720 RTP/AVP 0 8 18 2 96
a=rtpmap:2 G726-32/8000
a=rtpmap:96 telephone-event/8000          
a=fmtp:96 0-15

should this value of 96 bytes be 101?

R.

[raid_t]

Can anyone give a recommendation on what to check next?

Highly appreciated.

R.

[Evgeny Izetov]

Look in this document under the "rtp-nte" bullet point:

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html

I fixed a similar issue once by adding "asymmetric payload dtmf" under voice service voip > sip. Or you could try changing the default  payload type like the document says.

[raid_t]

Hi Evgeny,

I tried this but still not working!

Current section config:

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 fax protocol cisco
 sip
  bind media source-interface Dialer1
  registrar server expires max 1800 min 300
  asymmetric payload dtmf

In/Out dial peers have set:

dtmf-relay rtp-nte sip-kpml

Not sure if we're trying something which is not possible to work when using SCCP phones and ITSP does inband DTMF.

Thanks,

R.