Hello Everybody,
I have got tired & fedup for fixing the below common problem :
FXO Lines does not disconnect after playing IVR from CUE & ThirdParty Contact Center
I am having CME 9.1 connected with HQ through SIP Trunk. In HQ i am having CUCM 4.1 integrated with Altitude Contact Center Solution. Altitude is having IVR which plays the Welcome prompt for company.
I am having FXO Lines on CME. When i call from my mobile to the IVR of Altitude System & i drop the call from mobile, my FXO channel remain off-hook for 5 - 10 min & if i call to the CUE IVR (which is installed in my CME) the call disconnects in 20 - 30seconds
I searched alot on this issue & i got many solutions, i tried them as well but no luck for me. If any body can help me will be highly appreciated. following is my config:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/1.192
bind media source-interface GigabitEthernet0/1.192
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
!
voice class h323 1
h225 timeout tcp establish 3
h225 timeout setup 3
!
voice class dualtone-detect-params 1
freq-max-deviation 20
freq-max-delay 20
cadence-variation 20
!
voice class custom-cptone AR-custom
dualtone disconnect
frequency 416 416
cadence 400 320 240 520
!
voice translation-rule 2
rule 2 /.*\(5508\)/ /9903/
!
!
voice translation-profile 2
translate called 2
!
!
voice-port 0/2/1
translation-profile incoming 2
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone AR-custom
supervisory dualtone-detect-params 1
no battery-reversal
no vad
no comfort-noise
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 9033
description **** Voice Port for Etisal IVR ****
caller-id enable
!
dial-peer voice 973 voip
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
voice-class h323 1
dtmf-relay sip-notify
no vad
!
!
dial-peer voice 9033 voip
corlist outgoing call-national
destination-pattern 9033
session protocol sipv2
session target ipv4:192.168.0.10
voice-class codec 1
voice-class h323 1
dtmf-relay rtp-nte
no vad
!
!
