TRoute to Exchange UM for Voicemail

[nonny]

Hi guys

Have any of you successfully set up routing from SIP Server through to Exchange UM for Voicemail?  If so, could you please share your TServer options set against the Trunk DN and anything special you had to do?

Thanks

[mduran22]

Are you trying to go directly into voice mail for a specific extension without the ringing?

If that is the case you can adding opaque=app:voicemail to the end of the SIP URI. So your SIP URI would look something like this:

sip:user@domain.local;opaque=app:voicemail. I am actually setting this up to test now as well. I am going to try it two different ways, one with a force route directly if that's possible and another by setting this up as an extension on the switch and then targeting the extension in the strategy.

I'll post the results and anything else I try.

[nonny]

Yes that's pretty much exactly what I'm after.  I'll try the opaque option and see if that works.  Keep me updated with your progress too please!

[nonny]

I'm getting a 302 temporarily moved message back from the Exchange Server - which I understand is normal.  SIP Server should then be retrying the INVITE on port 5067.  I currently have two trunk DNs setup - one using port 5060 and one using port 5067 to the Exchange environment.  It doesn't appear that the re-INVITE is actually happening.  SIP is just responding to the 302 message with an ACK.

What do I need to enable or configure so that SIP Server receives the 302 message and then retries using the other Trunk DN on port 5067?

[cavagnaro]

Does UM indicates new Port?

[nonny]

UM replies with a 302 message and provides a new contact and to use port 5067 instead.

[cavagnaro]

And Sip server acknowledges it? Follow that sip conversation, may be some details there on why it is not doing.
As you don't share logs we are.blind