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CUCM | Cisco Collaboration

CME Integration with Cisco Unity

CME can be integrated with Cisco Unity, it is more or less as an integration with Call Manager except the fact that you must every setting in the CLI mode. Let’s see a kind of basic config:

!
telephony-service
voicemail 5555
!
ephone-dn 10
number 1234
call-forward busy 5555
call-forward noan 5555 timeout 10
ephone-dn 16
number 5566
mwi on
ephone-dn 17
number 5577
mwi off
ephone-dn 20
number 5555
name “VM-Port 1”
preference 0
no huntstop
ephone-dn 21
number 5555
name “VM-Port 2”
preference 1

no huntstop
ephone-dn 22
number 5555
name “VM-Port 3”
preference 2

no huntstop
ephone-dn 23
number 5555
name “VM-Port 4”
preference 3

!
ephone 20
vm-device-id “CISCOUM-VI1”
button 1:20
ephone 21
vm-device-id “CISCOUM-VI2”
button 1:21
ephone 22
vm-device-id “CISCOUM-VI3”
button 1:22
ephone 23
vm-device-id “CISCOUM-VI3”
button 1:23

So you can easily see that your config is trying to match the Unity port name and this is done with the command vm-device-id <<string>>. It helps also to register with Unity where you need the deviceId instead of a mac-address.

SRST : Quick Definition

Survivable Remote Site Telephony (SRST) is a feature which ensures that IP Phones can continue to function even if they are unable to communicate with Call Manager. During a failure, Cisco IP Phones register with the local SRST router which provides Call Processing and Control.

With the Connection Monitor Duration, Cisco IP Phones do not fail back immediately to ensure that the Call Manager in question is back online and is stable.

Here is also another presentation of Cisco SRST ( Cisco video)

Call Manager/SCCP forward to Unity

When a call is forwarded by Call Manager to Unity, the following information are included:

  • The called party’s extension
  • The calling party’s extension or external phone number
  • The reason that the call was forwarded to Unity (Busy, No-Answer, Inconditional)

So with these information, Unity is able to route the call to the correct mailbox.

Ringback issues

When you are working with H323 configurations, you can experience teh following ringtones issues , let’s see these issues and how to fix it quickly:

  • No Ringback tone on an IP Phone when calling a destination in the PSTN : This comes from a mistmatch between the Call Manager and the PSTN. It can be fixed with the following command progress_ind alert enable 8 under the POTS dial-peer.
  • No Ringback tone on a phone in the PSTN when calling an IP Phone: The problem is more similar with the previous one but it differs because it is coming from the other side as this time , the setup message received from the PSTN did not include a progress indicator (PI=0). Indeed, the gateway assumes the network is an ISDN end-to-end network so normally the tone must be handled by the PSTN but it is not an end-to-end call , this is why you have to update again the progress-indicator with the command progress_ind setup enable 3 which is setting the tone in-band.
  • No Ringback tone from the PSTN Phone which is forwarded from an IP Phone to another IP Phone : in this case, the issue is caused by the fact that the voice gateway is tearing down logical channel when you are transferring the call , so Audio is not sent. To fix this , you need then to change a parameter under the Call Manager setting ToSendH225UserInfoMsg and put it to True