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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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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