05-02-2011 04:05 AM - edited 03-16-2019 04:45 AM
Hi
I have ( PSTN router and CCM ver 8 and AA router for Auto Attendent) for customer , my question is
1) i have to configure AA in call manager express or Unity Express in AA Router ??? which one is better ?becasue already i have CCM version 8 so i dont need to configure CME , right ?
2) if i configured AA in CME as ACD configuration , what is the configuration i have to put in dial-peer to connect CME with CCM ver 8 becasue i want to configure all phones in CCM ver 8 ???
3) if i configured AA in unity EXpress , what is the configuration i have to put in dial-peer to connect unity-EXpress with CCM ver 8 ?
in actually i have a confused where i will configure AA in CME or unity-express and what is the dial-peer voip or pots i have to put and pointing to where ??
please if you have show run and description for dial-peer i will appritiate you for that
Note : All phones will configure in the CCM ver 8 not in CME
thanks and regards
05-02-2011 05:21 AM
Hello,
If you have CUCM then you don't need to configure CCME unless you want the phones associated with the CCME to be treated alone.
For the Unity express , if you have a CUCM , you have to confgure it to point to the call manager , when you configure the unity express , you configure two things on the CUCM , route point and CTI ports .
So the dial-peer should point to the destination pattern (cti route point) on the call manager .
Amer
05-02-2011 08:12 AM
Hi,
1 - If you have CUCM 8.0 you may need to make your CME router as SRST and configure the unity express connecting to CUCM by JTAPI
2 - here is the dial-peer configuration for unity express with CM 8
dial-peer voice 110 voip
description Voicemail in SRST mode
destination-pattern 603602952 this is the Voice mail number
session protocol sipv2
session target ipv4:10.134.170.10 this is the unity express IP
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 111 voip
description Autoattendant in SRST mode
destination-pattern 603602951 this is the AA number
session protocol sipv2
session target ipv4:10.134.170.10
dtmf-relay sip-notify
codec g711ulaw
!
dial-peer voice 112 voip
description Calls in non SRST mode for primary CUCM
destination-pattern 603602... this is starting extension in you site
session target ipv4:10.128.60.21 CUCM IP address
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
and you have to create a CTI Route Point for AA number in call manager and CTI Port, and also create a application user for JTAPI user and select all the CTI RP and CTI Ports under that user,also you have to enable the JTAPI in your CUE ISM and create user application for AA in CUE.
Regards,
05-02-2011 08:54 AM
thanks all for your helping , that's
right as i thought
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