Nortel to Unity via SIP trunk = no DTMF

[Andrew West]

Hey Everyone,

I have a functioning integration between CUCM 8.6 and a Nortel CS2100 over a SIP trunk.

Basic flow is like this:

CS2100 --- SIP TRUNK --- CUBE --- CUCM 8.6 + Unity Connection

We had to configure various outbound dial-peers so that when dialing anything on the Nortel side from a Cisco phone DTMF was processed and this works great. Here is an example:

dial-peer voice 12345 voip

description calls to XXXXX

destination-pattern 12345

session protocol sipv2

session target ipv4:1.1.1.1

incoming called-number 12345

dtmf-relay rtp-nte

codec g711ulaw

no vad

So basically dialing 12345 from a Cisco phone routes it to the session target which is an Auto-Attendant on the Nortel from there I can enter DTMF without issue.

Now I have a scenario were I am integrating Nortel phones into Unity and the first step is get the same call flow going the other way so my question is can I do an inbound dial-peer through the CUBE for DTMF normalization and make the session target Unity?

[fb_webuser]

On a CUBE/IOS gateway, each call uses two dial-peers, one for the incoming call leg and one for the outgoing call leg. You can figure out which dial-peer is used for the incoming call leg using the 'debug voip ccapi inout' debug. If you don't already have one matching then create a new incoming dial-peer. Here's a sample incoming dial-peer:

dial-peer voice 1 voip

description Incoming Dial Peer from SIP Server

session protocol sipv2

incoming called-number .

voice-class codec 1 OR codec g729r8 OR codec g711ulaw

dtmf-relay rtp-nte

If the called number are coming in with a specific prefix like an 8 then change the incoming called number to 8.

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Posted by WebUser Asad Raza from Cisco Support Community App

[fb_webuser]

Then you will need an outgoing dial-peer pointing to Unity

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Posted by WebUser Asad Raza from Cisco Support Community App

[Andrew West]

Hi,

That makes sense so this is the inbound DP i created , does it look right?

dial-peer voice 12345 voip

description Inbound test

session protocol sipv2

incoming called-number 12345

dtmf-relay rtp-nte

codec g711ulaw

no vad

[khayes1984]

Don't forget your session target.

Sent from Cisco Technical Support iPad App

[fb_webuser]

@westa@chop.edu

Yes this looks perfect. You don't need a session target for the incoming dial-peer.

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Posted by WebUser Asad Raza from Cisco Support Community App