05-06-2012 01:35 PM - edited 03-16-2019 11:00 AM
Guys,
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?
05-06-2012 02:00 PM
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.
HTH,
Chris
05-06-2012 02:09 PM
Chris,
+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
05-06-2012 02:18 PM
I have seen my answer here...
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
05-06-2012 03:37 PM
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.
HTH,
Chris
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