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CUE Auto Attendant not recognizing any digit presses, unable to route calls from outside callers.

fgramos01
Level 1
Level 1

It acts like its not getting any button presses.  I don't know if there is something wrong with our DTMF relay or if it is a telco issue that they are not sending dtmf signals to our router. Below is some parts of the current config on our router:





voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
session transport tcp
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g723r53
codec preference 4 g723r63
codec preference 5 g729br8


interface ISM0/0
ip unnumbered GigabitEthernet0/1
service-module ip address 172.16.4.130 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 172.16.4.1
!

control-plane
!
!
voice-port 0/1/0
trunk-group 4FXO
supervisory disconnect dualtone mid-call
connection plar 4100
description -- (072) 888-2085 -
!
voice-port 0/1/1
trunk-group 4FXO
supervisory disconnect dualtone mid-call
shutdown
!
voice-port 0/1/2
trunk-group 4FXO
supervisory disconnect dualtone mid-call
shutdown
!
voice-port 0/1/3
trunk-group 4FXO
supervisory disconnect dualtone mid-call
shutdown
!
!
!
!
!
!
mgcp profile default
!
sccp local GigabitEthernet0/1
sccp ccm 172.16.55.2 identifier 1 priority 1 version 6.0
sccp ccm 172.16.56.2 identifier 2 priority 2 version 6.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register reg1xcode
registration retries 15
registration timeout 8
keepalive retries 7
keepalive timeout 50
connect retries 5
connect interval 1200
switchback method graceful
switchback interval 120
!
dspfarm profile 1 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 10
associate application SCCP
!
dial-peer voice 4 pots
description ---To FXS Analog Phone---
destination-pattern 0113
!
dial-peer voice 5 voip
description --FXS to Manila PSTN--
destination-pattern *9[2-9]T
session target ipv4:172.16.55.2
!
dial-peer voice 7 voip
description --FXS to Reg2Tuguegarao PSTN--
destination-pattern *029T
session target ipv4:172.16.2.1
!
dial-peer voice 8 voip
description --FXS to Reg3Pampanga PSTN--
destination-pattern *039T
session target ipv4:172.16.3.1
!
dial-peer voice 9 voip
description --FXS to Reg5Pili PSTN--
destination-pattern *059T
session target ipv4:172.16.5.1
!
dial-peer voice 10 voip
description --FXS to Reg6Iloilo PSTN--
destination-pattern *069T
session target ipv4:172.16.6.1
!
dial-peer voice 11 voip
description --FXS to Reg7Cebu PSTN--
destination-pattern *079T
session target ipv4:172.16.7.1
!
dial-peer voice 12 voip
description --FXS to Reg8Tacloban PSTN--
destination-pattern *089T
session target ipv4:172.16.8.1
!
dial-peer voice 13 voip
description --FXS to Reg9aZamboanga PSTN--
destination-pattern *099T
session target ipv4:172.16.9.1
!
dial-peer voice 14 voip
description --FXS to Reg9bPagadian PSTN--
destination-pattern *049T
session target ipv4:172.16.9.129
!
dial-peer voice 15 voip
description --FXS to Reg10CDO PSTN--
destination-pattern *109T
session target ipv4:172.16.10.1
!
dial-peer voice 16 voip
description --FXS to Reg11Davao PSTN--
destination-pattern *119T
session target ipv4:172.16.11.1
!
dial-peer voice 17 voip
description --FXS to Reg12Koronadal PSTN--
destination-pattern *129T
session target ipv4:172.16.12.1
!
dial-peer voice 18 voip
description --FXS to Reg13Butuan PSTN--
destination-pattern *139T
session target ipv4:172.16.13.1
!
dial-peer voice 19 voip
description --FXS to Reg14Baguio PSTN--
destination-pattern *149T
session target ipv4:172.16.14.1
!
dial-peer voice 20 voip
description -- FXS to FXS --
destination-pattern [0-1]...
session target ipv4:172.16.55.2
voice-class codec 1
ip qos dscp cs3 signaling
!
dial-peer voice 21 voip
description --FXS to IP Phone--
destination-pattern [2-5]...
session target ipv4:172.16.55.2
ip qos dscp cs3 signaling
!
dial-peer voice 22 pots
trunkgroup 4FXO
destination-pattern 9.......
forward-digits 7
!
dial-peer voice 23 pots
trunkgroup 4FXO
description --From Remote Site to Local PSTN--
destination-pattern *019T
forward-digits 7
!
dial-peer voice 51 voip
description ----- FXS to IP Phone -----
preference 2
destination-pattern [2-5]...
session target ipv4:172.16.56.2
voice-class codec 1
ip qos dscp cs3 signaling
!
dial-peer voice 52 voip
description ----FXS to Remote FXS ---
preference 2
destination-pattern [0-1]...
session target ipv4:172.16.56.2
voice-class codec 1
ip qos dscp cs3 signaling
!
dial-peer voice 53 voip
description --FXS to Manila PSTN--
preference 2
destination-pattern *9[2-9]T
session target ipv4:172.16.56.2
!
dial-peer voice 54 voip
description --FXS to Reg1Batac PSTN--
destination-pattern *610T
session target ipv4:172.16.28.1
!
dial-peer voice 7000 voip
description ***** Reg1 La Union AUTO ATTENDANT *****
destination-pattern 4100
session protocol sipv2
session target ipv4:172.16.4.130
incoming called-number 9T
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 201 voip
!
!
!

4 Replies 4

Suresh Hudda
VIP Alumni
VIP Alumni

What is the call flow ? Does call comes inside from pstn using FXO ?

Try to use below dtmf relay in dial peer which is pointing to auto attendant.

dtmf-relay rtp-nte sip-notify

Suresh

when a customer tries to call our trunkline using either their landline or cellphone they will hear the auto attendant asking for a local number or if they don't know the local number they can dial 0 for our operator assistance. but even after they already pressed a digit multiple times the auto attendant will just say "are you still there?"

That's okay, I got that. But I'm asking call flow, you can describe as below.

mobile (pstn) -> fxo/pri -> voice gateway/cme -> cue.

Suresh

Hi. Sorry. yes call comes inside from pstn using fxo.

Already tried to use dtmf-relay rtp-nte sip-notify but still not working.

Thank you