Solved: Unable to select the call inputs in Cisco IP phone during a call

sharathpk0912 · ‎04-11-2016

I am unable to select the caller inputs in Cisco IP phones during a call. For example, I make a call to cisco TAC, if I press key '1' to open a new case, IP phone doesn't take the input.

Please help me to resolve the issue

Ayodeji Okanlawon · ‎04-11-2016

In cases like this, there are a few things to consider

1. The mode of your phone and what dtmf type it supports. Please tell us what the phone model is

2.. The inbound dial-peer from CUCM to the gateway. Do you have the correct dtmf configured here?

Please share your sh run so we can see what dtmf method you have configured here. if your ip phone doesn't support inband dtmf, I suggest you configure the following in your inbound dial-peer

dial-peer voice x voip

session-pro sip

dtmf-relay sip-kpml rtp-nte

incoming called-num . or (whatever number you want to use)

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Ayodeji Okanlawon · ‎04-11-2016

What is your setup? What systems are you using? CUCM, CCME? What type of gateways are you also using? Please describe your call flow

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sharathpk0912 · ‎04-11-2016

Call flow

Cisco IP phone----> CUCM-----> Voice Gateway------>PRI.

We are using SIP protocol.

CUCM version 10.5

Ayodeji Okanlawon · ‎04-11-2016

In cases like this, there are a few things to consider

1. The mode of your phone and what dtmf type it supports. Please tell us what the phone model is

2.. The inbound dial-peer from CUCM to the gateway. Do you have the correct dtmf configured here?

Please share your sh run so we can see what dtmf method you have configured here. if your ip phone doesn't support inband dtmf, I suggest you configure the following in your inbound dial-peer

dial-peer voice x voip

session-pro sip

dtmf-relay sip-kpml rtp-nte

incoming called-num . or (whatever number you want to use)

Please rate all useful posts

sharathpk0912 · ‎04-11-2016

running configuration is attached.

Phone model 7821/7861.

We are facing the issue in PRI line.

sharathpk0912 · ‎04-13-2016

I added the command session protocol sip-v2 and dtmf-relay sip-kpml under dial-peer voice x voip. It worked.

Thanks for your support.