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Community Member

Auto Attendant issue

i have configured auto attendant on uc520.internally its working fine and transfering calls to operator.but when i call on PSTN no. of auto attendant then i can hear the promts. but it is not accepting the keys. pressing 0 or 1 do nothing.prompt keeps on playing.any one can help? Thank You

4 REPLIES
Cisco Employee

Re: Auto Attendant issue

Make sure DTMF relay is configured under the dial-peers

HTH

java

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HTH

java

if this helps, please rate

www.cisco.com/go/pdi
Community Member

Re: Auto Attendant issue

this is my dial peers and i am using cisco configuration assistant

port 0/3/3
!
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1001 voip
corlist outgoing call-local
description ** star code to SIP trunk (Generic SIP Trunk Provider) **
destination-pattern *..
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1003 voip
description ** Passthrough Inbound Calls from CUE **
translation-profile incoming SIP_Passthrough
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ABCDT
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1005 voip
description ** Passthrough Inbound MWI from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number A80T
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1021 voip
corlist outgoing call-national
description **CCA*Generic Locale*all calls**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 1
destination-pattern 9[1-9].T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1023 voip
corlist outgoing call-local
description **CCA*Generic Locale*all calls**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 9[1-9].T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1024 voip
corlist outgoing call-local
description **CCA*Generic Locale*all calls**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 1
destination-pattern 9[1-9].T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1025 voip
corlist outgoing call-local
description **CCA*Generic Locale*all calls**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 1
destination-pattern 9[1-9].T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 3002 voip
description SIP
translation-profile incoming SIP_Called_6
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 2225248[1-9]
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 59 pots
trunkgroup ALL_FXO
corlist outgoing call-local
description **CCA*Generic Locale*all calls**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[1-9].T
forward-digits all
no sip-register
!
dial-peer voice 1002 voip
corlist outgoing call-local
description ** star code to SIP trunk (Generic SIP Trunk Provider) **
preference 1
destination-pattern *..
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:10.200.200.243:5060
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1020 voip
corlist outgoing call-national
description **CCA*Generic Locale*all calls**
translation-profile outgoing PSTN_Outgoing
preference 2
destination-pattern 9[1-9].T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:10.200.200.243:5060
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 100
b2bua
voice-class sip outbound-proxy  ipv4:10.1.10.1
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2003 voip
description ** cue auto attendant PSTN number **
translation-profile outgoing AA_Profile
destination-pattern 22252480$
b2bua
voice-class sip outbound-proxy  ipv4:10.1.10.1
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1006 voip
description ** AA from SIP Trunk (Auto Attendant 1)**
translation-profile incoming AA_Profile
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 22252480
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!

Community Member

Re: Auto Attendant issue

This is the config which works for me in CME-CUE VM & AA situation:

!
dial-peer voice 200 voip
description "Dial-peer for VM"
destination-pattern
session protocol sipv2
session target ipv4:
dtmf-relay rtp-nte sip-notify
codec g711ulaw
no vad
!

!
dial-peer voice 201 voip
description "Dial-peer for AA"
destination-pattern
session protocol sipv2
session target ipv4:

dtmf-relay rtp-nte sip-notify
codec g711ulaw
no vad
!

Community Member

Re: Auto Attendant issue

i have tried different dtnf-relay settings but it didn't

work. i am using sip trunk is there anythning to do from sip provider end?

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