Do you mean that I have to configure JTAPI Parameters?

Hi Ioanna,

You don't need any Jtapi config for a CUE with CME deployment

There are no Partitions or CSS's in this type of setup, those are for

"full blown" CUCM only and do not exist in CME.

I would have a good look through the following Guide. it is excellent;

Cisco CallManager Express/Cisco Unity Express Configuration Example

http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a008037f2a9.shtml

When you Call Forward an IP phone it won't be to 8000 or 8001 as these are
the MWI On/Off numbers. You would Forward to the VM Pilot# as defined below

It sounds like you are part way there.

Enter the command voicemail phone-number in order to define the number that is dialed when the Messages button is pressed.

Router(config-telephony-service)#voicemail 2000 (2000 is just an example here)

Hope this helps!

Rob

I have this command in my configuration.

When I press the msgs button I hear busy tone.

I am going to read the documentation and if I still have problem I am going to ask again.

Thanks a lot!

Hi Ioanna,

Make sure to check this section carefully;

Configure Dial Peers for Cisco CallManager Express Phones to Call Cisco Unity Express Voicemail

A dedicated dial peer is required to facilitate communication between       Cisco CallManager Express and Cisco Unity Express. The destination-pattern       under the dial peer assigns a set of phone numbers dedicated to applications       such as Cisco Unity Express. One thing that you need to consider is that the       voicemail number must be within the destination-pattern range of the dial peer,       and the ephone-dns must be configured for call forwarding to the voicemail       number which triggers the use of the dial peer. The Cisco Unity Express dial       peer must use Session Initiation Protocol (SIP) for call sessions and for the       translation of dual tone multifrequency (DTMF) tones into SIP notify messages.       A voice activity detection (VAD) must be switched off and a codec must be       designated.

1. Enter the command enable in order to             enter the EXEC mode.

   Router>enable
   

2. Enter the command configure terminal in             order to enter the configuration mode.

   Router#configure terminal
   

3. Enter the command dial-peer voice tag             voip in order to define a dial peer with a method of voice             encapsulation.

   Router(config)#dial-peer voice 2 voip
   

4. Enter the command destination-pattern             string in order to specify the prefix or the full E.164 number to             be used for the dial peer.

   Router(config-dial-peer)#destination-pattern 2...
   

5. Enter the command session protocol sipv2 in order to specify SIP as the session protocol for calls between host Cisco             CallManager Express router and Cisco Unity Express module.

   Router(config-dial-peer)#session protocol sipv2
   

6. Enter the command session target ipv4:destination-ip             address in order to designate a network-specific address to             receive calls from a voice over IP dial peer.

   Router(config-dial-peer)#session target ipv4:172.22.1.155
   

7. Enter the command dtmf-relay sip-notify in order to forward DTMF tones using SIP notify messages.

   Router(config-dial-peer)#dtmf-relay sip-notify
   

8. Enter the command codec codec in order             to specify the voice codec rate of speech for a dial peer.

   Router(config-dial-peer)#codec g711ulaw
   

   Note: Always ensure that there is no codec mismatch as any mismatch                 might result in inbound and outbound call failure.

9. Enter the command no vad in order to             disable VAD for the calls that use a particular dial peer.

   Router(config-dial-peer)#no vad
   

10. Enter the command exit in order to             finish the configuration.

    Router(config-dial-peer)#end
    

Cheers!

Rob

I tried this, but I still have problem.

When I am calling the 110 extension, the phone is ringing, after 10 seconds the phone stops ringing but I still hear the calling tone.

When I press the msg button I have busy tone and unknown number.

I am sure that is something is wrong in my configuration, but I don't know what it is

interface Service-Engine0/0
ip unnumbered FastEthernet0/0
service-module ip address 192.168.3.71 255.255.255.0
service-module ip default-gateway 192.168.3.1

!

!

ip route 192.168.3.71 255.255.255.255 Service-Engine0/0

!

!

dial-peer voice 300 voip
destination-pattern 2..
session protocol sipv2
session target ipv4:192.168.3.71
dtmf-relay sip-notify
codec g711ulaw
no vad

!

!

telephony-service

voicemail 200

!

!

ephone-dn  11  dual-line
number 110
call-forward busy 200
call-forward noan 200 timeout 10
corlist incoming call-full
corlist outgoing call-full

!

!

ephone-dn  70
number 8000...
mwi on
!
!
ephone-dn  71
number 8001...
mwi off

Is this configuration ok???

In the sh dial-peer voice 300 command the output is:

Admin state is up, Operation state is up,

DTMF Relay = enabled,

VAD = disabled, Poor QOV Trap = disabled,

codec = g711ulaw

Last Disconnect Cause is "26  ",

Last Disconnect Text is "network out of order (38)"