We use sip-kpml. Doesn't NTE always invoke an MTP because it's inband?

-----Original Message-----
From: cisco-voip [mailto:[email protected]] On Behalf Of Brian 
Meade (brmeade)
Sent: 04 February 2014 2:47 PM
To: Pawlowski, Adam; [email protected]
Subject: Re: [cisco-voip] SIP peers to UCM and DTMF

Adam,

Some of the phone models don't play the RFC 2833 inband DTMF tones audibly so 
that may be what you're seeing.  You'll probably need a packet capture to 
ensure the RTP events are being sent to the phone.

You shouldn't need to use MTP Required as an MTP will be dynamically pulled in 
for a DTMF mismatch.

Brian

-----Original Message-----
From: cisco-voip [mailto:[email protected]] On Behalf Of 
Pawlowski, Adam
Sent: Tuesday, February 04, 2014 4:29 PM
To: [email protected]
Subject: [cisco-voip] SIP peers to UCM and DTMF

Sorry if this one has been asked recently, but, what's the recommended 
configuration of the peers/trunk for SIP between an ISR gateway and a UCM 
(v9.1)? 

I have it set as rtp-nte which appears to work from the desk set -> pstn with 
no trouble, and PSTN inbound to Unity Connection or UCCX. I can see the 
messages generated in debug when I hit DTMF from the PSTN towards the desk set 
but no tone/beep is generated there. As well, I read that the older phones do 
not support rtp-nte, so I would need to have MTP required checked on the trunk 
for all calls? 

Just trying to make sure I have this all straight. 

Thanks,

Adam P
SUNYAB

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