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[–]flying_scrunt 4 points5 points  (2 children)

Make sure on the dial peer you have: dtmf-relay rtp-nte

[–][deleted] -1 points0 points  (1 child)

Ok, but it's not possible on the pots dial peer. Do I need to make another dial peer with incoming called number and not point it anywhere?

[–]flying_scrunt 2 points3 points  (0 children)

Ok, but it's not possible on the pots dial peer. Do I need to make another dial peer with incoming called number and not point it anywhere?

You should have a voip dial peer, for the call leg between your PBX and the CUBE, which uses that sip trunk. On the voip dial peer you can add rtp-nte for your dtmf relay. I would guess that you have an outbound dial-peer for your ISDN/PSTN side.

dial-peer voice 100 voip

  • description calls routed to your PBX
  • destination pattern <insert pattern here>
  • session-target ipv4:<IP of PBX>
  • dtmf-relay rtp-nte
  • session protocol sipv2
  • yadda yadda

side note: typically you would want 2 dial peers, at least, for for a given call (an inbound and outbound). Technically in some situations a single dial peer will match for both call legs, but I would try to avoid that as it's not flexible at all.

second side note: if you don't specify any dtmf-relay methods, DTMF tones will simply be left in the audio stream. I believe a DSP would convert NTE DTMF packets into tones in the audio stream for outbound....and the inverse for inbound.....I think.

[–]IDontDoStorage 1 point2 points  (0 children)

and I don't understand if I should make another dial-peer for the incoming call leg on the CUBE, in which I should set the DTMF type.

So you currently have no dial-peer for the inbound leg? Yes you will need one. Otherwise, you'll just be talking to our friend dial peer 0. And as /u/flying_scrunt says you'll probably need dtmf-relay rtp-nte as well.