DTMF Digits not getitng recognised

[gaurav_chawla1]

Hi,

We recently migrated from CME to CUCM Environment. Problem is when we call any number( mostly an IVR application ) , the call gets connected however when prompted to press any extension, phones doesn't seem to recognize the digits pressed and nothing happens further.

Call Flow :

Phones >> CUCM ( via SIP Trunk ) >> Gateway >>> PSTN

Gateway is H.323 and we are using an ISDN PRI to make outgoing calls.

Any help would be highly appreciated..!!

[Manish Gogna]

Hi Gaurav,

There is a very good doc on dtmf issues

https://supportforums.cisco.com/document/144711/understanding-dtmf-negotiation-and-troubleshooting-sip-trunks

Most likely it could be a failure of MTP allocation if there is a DTMF mismatch in the config of gateway dial peer and SIP trunk. Try checking MTP on SIP trunk and ensure proper MTP resources are assigned in the MRGL to begin with.

Manish

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[gaurav_chawla1]

Hi Manish,

Thanks for providing the link. It's indeed a very useful doc regarding dtmf.

DTMF is set to RFC 2833 on CUCM SIP trunk and rtp-nte on gateway. Checking the mtp allocation as suggested.

[Mohammed al Baqari]

Hi,

In your dialpeers config change the DTMF relay to sip-notify followed by sip-kpml. I agree with Manish that most probably it is DTMF RTP-NTE problem.

There are two ways of fixing this:

1. Use out-of-band DTMF relay such as SIP-NOTIFY , SIP-KPML , H245-ALPA

2. Use MTP with RTP-NTE

[avinsrid89]

Hi Gaurav,

Since CUCM to gateway is via SIP Trunk, mostly the dtmf is going to go via RTP-NTE (RFC-101) which is in band DTMF.

An MTP is needed only if you convert in band to out of band dtmf for eg. 

device <--- SIP RTP-NTE(In band) --> gateway <--- H.245 Signal(out of band) --->

MTP on the gateway would then convert the tone.

From your scenario since the router is connecting to PSTN, the voice gateway is self capable of converting the digits that come from CUCM and send to PSTN.

1) I would suggest, "debug voice ccapi in out" to see if the DTMF digits enter the Ccapi layer of router. If it does, then most likely these digits are going to the PSTN. Your issue would then be in PSTN side.

2) Try the debug for a call and press digits on the phone to see if the relay is there. If we don't see it, then we take CCM traces and check how the digits are getting lost. 

This is the best way to troubleshoot such issues.

Kindly rate useful posts!

~Avinash

[gaurav_chawla1]

" Media Termination Point Required " is unchecked on the SIP Trunk and NO MTP resources are assigned in the MRGL since no MTP Profile is configured on the gateway except for conference and transcoding.

[Mohammed al Baqari]

The configure sip-notify as dtmf relay on your dialpeers 

[avinsrid89]

Hi Gaurav,

Since CUCM to gateway is via SIP Trunk, mostly the dtmf is going to go via RTP-NTE (RFC-101) which is in band DTMF.

An MTP is needed only if you convert in band to out of band dtmf for eg. 

device <--- SIP RTP-NTE(In band) --> gateway <--- H.245 Signal(out of band) --->

MTP on the gateway would then convert the tone.

From your scenario since the router is connecting to PSTN, the voice gateway is self capable of converting the digits that come from CUCM and send to PSTN.

1) I would suggest, "debug voice ccapi in out" to see if the DTMF digits enter the Ccapi layer of router. If it does, then most likely these digits are going to the PSTN. Your issue would then be in PSTN side.

2) Try the debug for a call and press digits on the phone to see if the relay is there. If we don't see it, then we take CCM traces and check how the digits are getting lost. 

This is the best way to troubleshoot such issues.

Kindly rate useful posts!

~Avinash