Exchange 2010 UM integration - strange issue from local CME phones

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Feb 28th, 2010
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We took the plunge on Exchange 2010 and UM integration.  We have a PRI and have NO problems with speaking commands from calls originating over the PRI.

However - one of my dial peers just will not send the audio from the mic on the phone to the exchange server when calling from inside (using a Cisco phone).  The internal phones can send DTMF succesfully.  But if you tried to record your name greeting or speak a command to UM - no go.  The Cisco phones just cant send audio out that particular dial peer?


What would cause PRI calls to have two-way communications with audio but internal calls to have only one-way (receive audio) but not send audio to a SIP dial peer?  I should say that maybe after 5 attempts - the audio from a Cisco phone does make it through to the UM server - but that is very unreliable for internal calls.


Some of our global settings and dial-peers:


voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
  h225 timeout ntf 100
  h225 display-ie ccm-compatible
sip
  no update-callerid


dial-peer voice 2012 voip
destination-pattern 560
b2bua
session protocol sipv2
session target ipv4:192.168.1.25
session transport tcp
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad
!

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Steven Smith Sun, 02/28/2010 - 17:53
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It could be a difference in dial-peer that is being hit on the calls.



Can you provide..

debug ccsip message

debug voip ccapi inout

show run (remove passwords)




Can you do this for a PRI call, bad IP Phone call, and a good IP Phone call.

VINH DOAN Tue, 04/06/2010 - 09:48
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Rob,


I am also doing a SIP integration with Exchange 2010. Do you have a sample config that I can look at? This will be my first time doing it. Thank you.


Vince Doan

Nexus



my email is:  vince.doan@nexusis.com

robby.hill Tue, 04/06/2010 - 10:45
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What is not apparent from the Microsoft documentation - but something I found out by debuging my SIP calls to my Microsoft UM server - is that the Auto Attendant with Microsoft UM is on a different port number.


Rather then reinvent the wheel, I just edited the target for my original Cisco voicemail attendant to be the Microsoft UM box on the correct port number.  Thatlooks like this:


dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 505
b2bua
session protocol sipv2
session target ipv4:192.168.1.25:5065
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad


Of course - for my voicemail target I did the same thing:


dial-peer voice 2012 voip
destination-pattern 560
b2bua
session protocol sipv2
session target ipv4:192.168.1.25:5065
session transport tcp
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad


These work like a charm.  I created a DID for my UM for employees to check their voicemails remotely.  I just changed my voicemail button setup (under voice in the CME config) to the DID number that is for my number that we setup for checking messages remotely.

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