ā04-11-2016 01:56 AM - edited ā03-17-2019 06:31 AM
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
Solved! Go to Solution.
ā04-11-2016 08:15 AM
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)
ā04-11-2016 02:04 AM
What is your setup? What systems are you using? CUCM, CCME? What type of gateways are you also using? Please describe your call flow
ā04-11-2016 07:10 AM
Call flow
Cisco IP phone----> CUCM-----> Voice Gateway------>PRI.
We are using SIP protocol.
CUCM version 10.5
ā04-11-2016 08:15 AM
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)
ā04-11-2016 11:07 PM
ā04-13-2016 12:48 AM
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.
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