[cisco-voip] SIP peers to UCM and DTMF

[Eric Pedersen]

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

-----Original Message-----
From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Meade (brmeade)
Sent: 04 February 2014 2:47 PM
To: Pawlowski, Adam; cisco-voip at puck.nether.net
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:cisco-voip-bounces at puck.nether.net] On Behalf Of Pawlowski, Adam
Sent: Tuesday, February 04, 2014 4:29 PM
To: cisco-voip at puck.nether.net
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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