Cisco Unity ignoring external input

[eklock]

When trying to press * or 4 during cisco greeting from an external caller it is getting ignored. Internal works fine.

I read somewhere about the voice gateway dmtf settings but I believe I have all the correct info.

Here is some of the running config on the gateway.

control-plane
!
!
voice-port 0/0/0:23
!
voice-port 0/1/0
timing hookflash-out 50
timing guard-out 1000
!
voice-port 0/1/1
!
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server cucm01
ccm-manager config
!
mgcp
mgcp call-agent 192.168.100.10 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
!
mgcp profile default
!
!
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
!
dial-peer voice 100 voip
description 10 digits to CUCM
preference 1
destination-pattern 663[1-4]
session protocol sipv2
session target ipv4:192.168.100.10
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.100
voice-class sip bind media source-interface GigabitEthernet0/0.100
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
no vad
!
dial-peer voice 101 voip
description 10 digits to CUCM
preference 1
destination-pattern 5924
session protocol sipv2
session target ipv4:192.168.100.10
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.100
voice-class sip bind media source-interface GigabitEthernet0/0.100
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
no vad
!
dial-peer voice 102 voip
description 10 digits to CUCM
preference 1
destination-pattern 8820
session protocol sipv2
session target ipv4:192.168.100.10
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.100
voice-class sip bind media source-interface GigabitEthernet0/0.100
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
no vad
!
dial-peer voice 200 voip
description Outbound to Siptrunk
destination-pattern [2-9]..[2-9]......$
session protocol sipv2
session target ipv4:192.168.101.15
voice-class codec 1
voice-class sip early-offer forced
voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 201 voip
description Outbound to Siptrunk
destination-pattern 1[2-9]..[2-9]......$
session protocol sipv2
session target ipv4:192.168.101.15
voice-class codec 1
voice-class sip early-offer forced
voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 202 voip
description Outbound to Siptrunk
destination-pattern [2-9]......$
session protocol sipv2
session target ipv4:192.168.101.15
voice-class codec 1
voice-class sip early-offer forced
voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 911 voip
description 911 to Midco
destination-pattern [49]11
session protocol sipv2
session target ipv4:192.168.101.15
voice-class codec 1
voice-class sip early-offer forced
voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711ulaw
no vad
!
!
sip-ua
retry invite 2
!
!
!
gatekeeper
shutdown
!
!
!
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
privilege level 15
login local
transport input telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
!
scheduler allocate 20000 1000
end

Thanks!

[Jonathan Schulenberg]

I’d start by pairing DTMF back; try this:

dtmf-relay rtp-nte sip-kpml

[Roger Kallberg]

This type of issue is definitely related to DTMF. Your configuration on the dial peers for this is including options that are irrelevant as they are for H.323. Change your DTMF relay to this “dtmf-relay rtp-nte sip-kpml”, possibly you’ll might need to add sip-notify as well, at least for the service provider dial peers. It depends on the requirements from your service provider.

[eklock]

I switched all the dial peers to jus tthe dtmf-relay rtp-nte sip-kpml and still it won't accept any external input. I then tried to add the sip-notify but same result. Really strange. The internal * command works internally calling from one phone to another. Just not from any external phones. Any other settings I should be checking?

Thanks!

[Roger Kallberg]

What type of integration do you use for CUC, SIP or SCCP?

[eklock]

Most of the phones are using SIP a couple are still on SCCP. That might be some of the issue. Originally it was all SCCP and a while back we changed the phones over to SIP. Not much of an expert on the Cisco phone side of things so any input would be much appreciated. Any place I could look in CUCM that could point me in the right direction.

Thanks!

[eklock]

I did find the port group to set up as SCCP integration in the unity connection settings.

[Roger Kallberg]

If you look on the port group what DTMF variants do you have set?

[eklock]

I don't see many settings for DTMF on the web interface besides the screenshot below:

[Roger Kallberg]

Have a look at this document. Troubleshooting Guide for Cisco Unity Connection Release 11.x 

Don’t pay to much attention to the part about DTMF relay on the dial peers as that is outlined for a H.323 configuration.

[eklock]

Sorry, I should have mentioned that this is a very old system running 10.5 cuc. I don't have the DTMF KPML and Use DTMF RFC 2833 check boxes under advanced settings.

[Jonathan Schulenberg]

One possible cause would be CUCM invoking an MTP - especially an IPVMSA SW MTP that lacks DTMF interworking support. I suggest checking the SIP trunk to ensure MTP Required is not checked. If it is, be careful to test the system thoroughly after disabling it and resetting the trunk; someone may have checked that as a bandaid instead of properly fixing some other issue.

[eklock]

I unchecked MTP over the weekend to test but still no external user input was recognized. I left that unchecked while changing around and trying a few different dtmf-relay settings but nothing was working. I set the mtp back on for now. Still haven't found a solution yet. It is an old system. 10.5 version running with a 2901 router.