I have two sites with CUCME 7.0 routers. Site A has a PRI for outbound calls and a T1 connection to Site B. Site B forwards its outbound calls to Site A to go out the PRI.
When Site B makes an outbound call the call connects, but after that it doesn't forward any DTMF tones. You can hear the tones on the handset but it doesn't forward the tones outbound.
Site A does not have this problem.
I've tried a number of different dtmf-relay configs but nothing seems to help.
Here's the current dial peer config:
dial-peer voice 152 voip
description <<11 Digit Long Distance Numbers>>
session protocol sipv2
session target ipv4:192.168.150.1
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I would shy away from sip-notify. It's primary use is for CUE, and that's about it.
I would use rtp-nte for SIP configuration.
Otherwise you are probably hitting dial peer 0. Do a 'debug voip dialpeer'. Be aware the the outgoing dial peer on one router must have the DTMF, as well as the INCOMING dial peer on the other router. This is most likely your problem, as incoming dial peers are very commonly overlooked.
Try adding an 'incoming called-number .' to your outgoing dial peer on both sides and see if you get different results.
I have tried rtp-nte and h245-alphanumeric without any luck. I have not checked to make sure the dtmf line is also part of the incoming dial peer as well. I'll also try adding the "incoming called-number ." to the outbound dial peer. One question on that though, won't that conflict with the existing incoming dial peer for inbound calls on the PRI?
Hi, incoming called-number doesn't do anything for outgoing DP, and if trunking is working already, you don't need it at all. But, you need incoming-calling number on the incoming voip DP to make sure the dtmf relay agrees on both sides.
Also, recommend you try H.323.
Thanks to all! I removed the "session protocol sipv2" from the outboud dial peer and it's now working.
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