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 ! ... View more