Issue the incoming call to IVR with cucm centralized

[Hary_CsC]

Haii,

I have a problem with the incoming call, so when called 02126507525 from 27301 (IVR) the number could not be reached.

*cucm centralized

Configuration VG as follows:
!
voice service voip
mode border-element license capacity 30
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip handle-replaces
supplementary-service media-renegotiate
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through none
sip
registrar server expires max 600 min 60
early-offer forced
midcall-signaling passthru
!
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 5
no call preserve
!
voice translation-rule 2
rule 1 /02126507525/ /27301/
!
voice translation-profile INCOMING-SIP-TELKOM
translate called 2
!
dial-peer voice 2001 voip
description ** via SIP Telkom **
translation-profile incoming INCOMING-SIP-TELKOM
session protocol sipv2
session target ipv4:x.x.x.x
incoming called-number ^02126507525$
voice-class codec 1
voice-class sip transport switch udp tcp
voice-class sip early-offer forced
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
no vad
!
dial-peer voice 105 voip
description *** Outbound LAN Side Dial-Peer (INCOMING CUCM Publisher) ***
destination-pattern 3....
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip transport switch udp tcp
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 106 voip
description *** Outbound LAN Side Dial-Peer (INCOMING CUCM Subscriber-1) ***
destination-pattern 3....
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip transport switch udp tcp
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!

Please advise.

Thx

[Roger Kallberg]

There is nothing that routes the resulting translated number to CM in your posted configuration. Correct that and it would likely work. Assuming you have the needed configuration in-place in CM obviously.

[Roger Kallberg]

Another observation is that your dial peer to the subscriber is H323. Is that intentional?

[Hary_CsC]

Hai Roger Kallberg 

Have a nice day,
i've routed dial-peer to cucm. 
dial-peer voice 2001 voip
description ** via SIP Telkom **
translation-profile incoming INCOMING-SIP-TELKOM
session target ipv4:x.x.x.x

x.x.x.x is CUCM IP, so x is't published.

I don't use voice class h323 on Ip phone device or dial-peer, why you mention it ?
Please advise.

Thx

[Roger Kallberg]

Your dial peers that have a destination pattern is different.

dial-peer voice 105 voip
description *** Outbound LAN Side Dial-Peer (INCOMING CUCM Publisher) ***
destination-pattern 3....
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip transport switch udp tcp
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad

This has session protocol sipv2, that makes it a SIP dial peer.

dial-peer voice 106 voip
description *** Outbound LAN Side Dial-Peer (INCOMING CUCM Subscriber-1) ***
destination-pattern 3....
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip transport switch udp tcp
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad

This does not have any session protocol defined, so it will use the default and that's H.323.

Have you done a debug ccsip message to see that the call is sent to CM? I personally don't like the use of dial peer as both inbound and outbound, but it for sure should work with proper configuration. Do you see that the call is hitting the dial peer 2001 and then using the same in the outbound direction?