DTMF is not working from PSTN to CUC

[ali nasser]

Dears,

I'm having an issue with DTMF from the PSTN to CUC. I tried all the changes as per the community, but no luck. 

The integration is SIP between the CUC and CUCM.

Call Flow:

ISP---->CUBE(SIP)-----> CUCM(SIP)------>CUC (SIP).

here are the dial-peers;

dial-peer voice 110 voip
description Calls in primary CUCM
destination-pattern 2...
session protocol sipv2
session target ipv4:172.16.65.12
voice-class codec 1
dtmf-relay rtp-nte
no vad
!

dial-peer voice 11 voip
description outbound to STC
translation-profile outgoing PSTN-OUT
destination-pattern 905........
session protocol sipv2
session target dns:fmc.stc.com.sa:5060
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 21
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 1 voip
description inbound from STC
translation-profile incoming PSTN-IN
session protocol sipv2
session target ipv4:10.154.15.25
incoming called-number +9661T
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
no vad

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
localhost dns:fmc.stc.com.sa
midcall-signaling passthru
sip-profiles 21

BR

[Roger Kallberg]

Hi @ali nasser,
How has it gone with this? Have you tested with the suggested changes that I made a few days ago? Apart from them I would also recommend you to add this to you configuration.

voice service voip
 ip address trusted list
  ipv4 172.16.65.11
  ipv4 172.16.65.12
  ipv4 172.16.65.13
  ipv4 10.154.15.25
 no allow-connections h323 to h323 !not needed as you do SIP 2 SIP
 no allow-connections h323 to sip !not needed as you do SIP 2 SIP
 no allow-connections sip to h323 !not needed as you do SIP 2 SIP
 no h323 !not needed as you do not use H.323

[ali nasser]

Hi @Roger Kallberg 

Thanks for your email.

I'm waiting to have access to the device, it will be on Monday. i will update if this works. Also, the dtmf is not working even if you call a company with IVR.

BR

[Ibrahim AlQumairi]

Try to activate the "Media Termination Point Required" on the sip trunk that points to the CUBE.

[Roger Kallberg]

That should not be needed and if it is then it's IMHO a tell sign that something is wrongly configured.

[Carlo Poggiarelli]

Hi ,

Can you please activate a debug voip ccapi inout and a debug voip rtp sess name place a call to VM and post the result?

Thanks

Regards

Carlo