CME with Microsoft Lync/Mediation server Integration

Hello
please, we want to integrate our Cisco CME with microsoft Lyn/Mediation Server.
Please, what do i need to configure on the Cisco CME?
Thanks

I have config on cme side. But not Lync side(
sip-ua
no remote-party-id
voice service voip
allow-connections sip to sip
no supplementary-service sip handle-replaces
sip
bind control source-interface gi0/0 (this should be your sip interface, creating loopback and binding to SIP is best practice)
bind media source-interface gi0/0 (this should be your sip interface, creating loopback and binding to SIP is best practice)
voice class codec 20
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
dial-peer voice 11 voip (Incoming dial-peer from Lync)
b2bua
session protocol sipv2
session target ipv4:LyncIP
session transport tcp
incoming called-number .%
voice-class codec 20
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 21 voip (outgoing dial-peer to Lync)
destination-pattern 1... (this is Lync patterns dial-peer)
notify redirect ip2pots
session protocol sipv2
session target ipv4:LyncIP
session transport tcp
voice-class codec 20
dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711ulaw
no vad

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Thanks to all for yours support.
Today, we have done some test (no employee in office today
J) and we have solved the problems.
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http://technet.microsoft.com/en-us/library/jj205160.aspx
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Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a marked post does not actually answer your question. This can be beneficial to other community members reading the thread.
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I Turned off CAC and left Media Bypass on. I received 87% packet loss, although when listening to the voicemail, I heard no drops at all. Here are the stats:
Audio Stream (Caller -> Callee)
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    Avg. round trip: 0 ms Max. round trip: 0 ms
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Audio Stream (Callee -> Caller)
    Codec: PCMA Sample rate: 8000
    Audio FEC: False Bandwidth estimates: 0 Kbps
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    Avg. packet loss rate: 0.00 % Max. packet loss rate: 0.00 %
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    Audio Stream (Caller -> Callee)
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    Packet utilization: 1866
    Avg. packet loss rate: 0.00 % Max. packet loss rate: 0.00 %
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    Avg. network MOS degradation: 0.15 Max. network MOS degradation: 0.18
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I'm thinking it's a problem with the reporting feature since no visible/physical packet loss can be seen or heard. Our issue also lies with the fact that it's throwing of our call quality stats for the day because of the amount of "poor calls" going
to voicemail.
I would really love to figure this out. Any thoughts?
John K. Boslooper Unified Communication Engineer

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