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Voip dial peer and dtmf-relay query


There is a config that I am not quite sure about. Off the top of my head, I say its not right..but are there issues that this config may cause?

Here is it:

dial-peer voice 1 voip

  incoming called-number . 

dtmf-relay rtp-nte digit-drop h245-alphanumeric

++++ Note that its a h323-dial-peer with a sip dtmf relay type...+++

I believe the right thing is to have

session portocol sip v2 on that dial-peer or change the dtmf type?

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Chris Deren
Hall of Fame Master

So, do you need H323 or SIP dial peer? For SIP the most common config is :

dial-peer voice 1 voip

session protocol sipv2

dtmf-relay rtp-nte

no vad

with H323:

dial-peer voice 1 voip

dtmf-relay h245-alphanumeric

no vad

However keep in mind that there are few DTMF standards and the config for DTMF may vary depending on which one is used.




+5 for your time

I know the standards..I have just seen this on a gateway i am working on...and I am at a loss why its configured this way. I am wondering if this could create issues. I deally i would have it as you suggested which is what I wrote earlier too

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I have seen my answer here...

Configuring DTMF Relay Digit-Drop on an Cisco Unified Border Element with Cisco Unified Communications Manager

To avoid sending both in-band and out-of band tones to the outgoing leg when sending Cisco UBE calls in-band (rtp-nte) to out-of band (h245-alphanumeric), configure the dtmf-relay rtp-nte digit-drop command on the incoming SIP dial-peer. On the H.323 side configure either dtmf-relay h245-alphanumeric or dtmf-relay h245-signal command. This feature may also be used for H.323-to-SIP, and H.323-to-H.323 calls.

+++That dtmf type is only relevant when doing h323-SIP calls.+++ rtp-nte should not be on the h323 leg as expected

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Correct, how is this GW defined in CUCM is it H323 GW or a SIP trunk? Depening on which one it is either adjust the dtmf method or add sipv2 to the dial-peer.



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