Unity Connection Call Handler Input ignores from External Phones

[Slavi]

Hello,

for my Customer, i´ve created a Call Handler for Callnavigation with User Input.

When i dial the Call Handler from an internal Phone, the Caller Input is working fine. (For Example: press 1 to get redirected to Service Department).

But when i call the Call Handler from an external Phone, the Caller Input will be ignored.

What is wrong?

In my CUCM i´ve created a CTI Route Point with the Call handler Number and set Forward all to Voicemail.

Please help me.

Thanks a lot.

Greetings

[Abhay Singh Reyal]

If internally it is working fine then there must be no issue with the configuration of AA on Cisco Unity Connection. Externally if it is not working then it simply means it is not recognizing the DTMF input.
Provide the exact call flow for the calls coming in from external network. According to that flow will advise you to collect the debugs.
For Ex :- ITSP >> SIP >> CUBE >> SIP >> CUCM >> SIP >> CUC

Regards
Abhay

[Chris Deren]

In addition provide "show run" from your ingress voice gateway/CUBE to which the call arrives.

[Slavi]

Hi Chris,

thanks for your reply.

Ive added the sh run from the Voice GW.

voice-card 0
dsp services dspfarm
!
!
!
voice service voip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
fax-relay ans-disable
sip
bind control source-interface port-ch1
bind media source-interface port-ch1
no update-callerid
!
voice class codec 1
codec preference 1 g711alaw
!
voice class sip-profiles 1
response ANY sip-header Remote-Party-ID remove
!
voice class sip-options-keepalive 10
down-interval 60
!
!
!
!
voice translation-rule 1
rule 1 /^\(.*\)/ /+49\1/ type national national
rule 2 /^\(.*\)/ /+\1/ type international international
!
voice translation-rule 10
rule 1 /^610/ /9999/
rule 2 /^61\(1...$\)/ /\1/
rule 3 /^61\(2[0167]..$\)/ /\1/
rule 4 /^61\(22..$\)/ /\1/
rule 5 /^612\([3489]...$\)/ /\1/
rule 6 /^612\(5....$\)/ /\1/
!
voice translation-rule 11
rule 1 /^610/ /9999/
rule 3 /^61\(1...$\)/ /\1/
rule 4 /^61\(2[0167]..$\)/ /\1/
rule 5 /^61\(22..$\)/ /\1/
rule 6 /^612\([3489]...$\)/ /\1/
rule 7 /^612\(5....$\)/ /\1/
rule 16 /^61\(.*\)/ /\1/
!
voice translation-rule 100
rule 1 /^\+4911\(.$\)/ /11\1/
rule 2 /^\+49\(.*\)/ /0\1/
!
voice translation-rule 110
rule 1 /^\+\(.*\)/ /00\1/
!
voice translation-rule 200
rule 1 /^\+49\(.*\)/ /\1/ type any national
rule 2 /^\+\(.*\)/ /\1/ type any international
!
voice translation-rule 210
rule 1 /^\+49\(.*\)/ /49\1/ type any international
rule 2 /^\+\(.*\)/ /\1/ type any international
!
!
voice translation-profile VOICE_TRANS_INCOMING_PSTN
translate calling 1
translate called 10
!
voice translation-profile VOICE_TRANS_PSTN_TO_INTERNAT
translate calling 210
translate called 110
!
voice translation-profile VOICE_TRANS_PSTN_TO_NATIONAL
translate calling 200
translate called 100
!
!
!
license udi pid CISCO2911/K9 sn FCZ2011607D
hw-module pvdm 0/0
!
!

redundancy
!
!
!
!
!
controller E1 0/0/0
pri-group timeslots 1-31
!
controller E1 0/0/1
pri-group timeslots 1-31
!
!
!
!
!
!
!
!
!
!
!
interface Port-channel1
ip address 10.254.254.174 255.255.255.252
vlan-id dot1q 900
exit-vlan-config
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
channel-group 1
!
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
channel-group 1
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving T302 3000
isdn incoming-voice voice
isdn sending-complete
trunk-group TRUNK_GROUP_PSTN 1
no cdp enable
!
interface Serial0/0/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving T302 3000
isdn incoming-voice voice
isdn sending-complete
trunk-group TRUNK_GROUP_PSTN 1
no cdp enable
!
ip forward-protocol nd
!
no ip http server
ip http access-class 1
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
ip route 0.0.0.0 0.0.0.0 10.254.254.173
ip ssh authentication-retries 2
ip ssh version 2
ip scp server enable
!
logging host 141.46.10.2
logging host 141.46.10.3
logging host 141.46.10.7
!
!
control-plane
!
!
voice-port 0/0/0:15
disc_pi_off
bearer-cap 3100Hz
!
voice-port 0/0/1:15
disc_pi_off
bearer-cap 3100Hz
!
!
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 1 pots
description ***DEFAULT-INCOMING-DIALPEER***
incoming called-number .
direct-inward-dial
!
dial-peer voice 100 pots
translation-profile outgoing VOICE_TRANS_PSTN_TO_NATIONAL
destination-pattern +49
port 0/0/0:15
!
dial-peer voice 110 pots
translation-profile outgoing VOICE_TRANS_PSTN_TO_INTERNAT
destination-pattern +
port 0/0/0:15
!
dial-peer voice 11 voip
translation-profile outgoing VOICE_TRANS_INCOMING_PSTN
preference 1
destination-pattern 61T
session protocol sipv2
session target ipv4:172.31.100.2
session transport tcp tls
incoming called-number .
voice-class codec 1
voice-class sip url sip
voice-class sip srtp negotiate cisco
voice-class sip profiles 1
voice-class sip options-keepalive
dtmf-relay sip-notify rtp-nte
srtp fallback
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
dial-peer voice 21 voip
translation-profile outgoing VOICE_TRANS_INCOMING_PSTN
preference 2
destination-pattern 61T
progress_ind setup enable 3
session protocol sipv2
session target ipv4:172.31.200.2
session transport tcp tls
incoming called-number .
voice-class codec 1
voice-class sip url sip
voice-class sip srtp negotiate cisco
voice-class sip profiles 1
voice-class sip options-keepalive
dtmf-relay sip-notify rtp-nte
srtp fallback
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
dial-peer voice 101 pots
translation-profile outgoing VOICE_TRANS_PSTN_TO_NATIONAL
huntstop
preference 2
destination-pattern +49
port 0/0/1:15
!
dial-peer voice 111 pots
translation-profile outgoing VOICE_TRANS_PSTN_TO_INTERNAT
destination-pattern +
port 0/0/1:15
!
!
sip-ua
crypto signaling remote-addr 172.31.0.0 255.255.0.0 trustpoint VOICE-GATE-ZI strict-cipher
!
!
!
gatekeeper
shutdown
!
!
!
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
access-class 199 in
transport input ssh
line vty 5 15
access-class 199 in
transport input ssh
!
scheduler allocate 20000 1000
ntp server 141.46.8.1
!
end

[Chris Deren · Accepted solution]

On your voip dial peer change:

dtmf-relay sip-notify rtp-nte

to 

dtmf-relay rtp-nte sip-kpml

Obviously, after making this change test different DTMF scenarios besides voicemail test, i.e. outbound DTMF, inbound to other system you may have (contact center, etc.).

[Slavi]

Hello Chris,

your Solution is working fine.

Thanks a lot for your help! :)

Greetings,

Slavi