Scenario:
Call Manager (3.1.3aspF) with Unity Integration (3.1.2b)
Centralized call processing | Router= C4224 w/2FXO cards 1-wic1dsuT1
When the data T1 is up the calls are plarred to a CTI route point on the call manager which then sends the call to unity for schedule checking, then if the schedule is open its sent back to call manager and then out to the correct gateway to ring the phone.
When the Data T1 goes down and SRST kicks in there is no way to translate the plarred number on the voice ports. To ring a DN on the call-manager-fallback dial-peers. I need the equivalent of hairpinning which would use dial-peer 21 when the Data T1 is up and dial-peer 81 when its down... Doesn't work.. Anyone see what i am doing wrong???
So say the config is like this:
voice-port 3/0
connection plar 8590
!
voice-port 3/1
connection plar 8590
! This is for the 1st part of the outgoing call leg to the pstn
dial-peer voice 20 voip
description Outbound call leg for Internal DN's 4950-4959
preference 1
destination-pattern 495.
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:1.2.3.4
dtmf-relay h245-alphanumeric
ip qos dscp cs5 media
no vad
! This is the second call leg for outgoing calls
dial-peer voice 92 pots
description First Analog DP matched for Outbound/Originate call legs Local
preference 1
destination-pattern 9T
port 3/1
! For routing the plarred number to Call Mgr.
dial-peer voice 21 voip
description Outbound call leg for Unity time check on incoming calls
preference 1
destination-pattern 8590
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:161.137.24.11
dtmf-relay h245-alphanumeric
ip qos dscp cs5 media
no vad
! This is to avoid using the default dial-peer (pid:0) 1st call leg incoming calls
dial-peer voice 83 pots
description Dum Dial-peer for calls with no ANI/CLID
incoming called-number .
destination-pattern .
port 3/0
! This dial-peer creates a hunt group and goes want it to become active when the Data T1 goes down...
1st try i used:
dial-peer voice 81 pots
preference 2
destination-pattern 8590
translate-outgoing called 10
port 3/0
2nd try i used:
dial-peer voice 81 voip
preference 2
destination-pattern 8590
progress_ind setup enable 3
translate-outgoing called 10
voice-class codec 1
voice-class h323 1
session target ipv4:161.137.24.11
dtmf-relay h245-alphanumeric
ip qos dscp cs5 media
no vad
! Translation rule for dial-peer 81
translation-rule 10
Rule 0 8590 3807570 unknown network
