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
!
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.
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
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.