Exchange Online / Office 365 Unified Messaging with CUCM via CUBE

[jonathan.dixon]

Hi All,

Wondering if anyone has tried using CUBE running on an ISR (eg. 2901-VSEC/CUBE bundle) as a SBC to connect CUCM 10.5 to Exchange in the cloud for Unified Messaging (voicemail)?  Cisco have a guide for doing this on an ASR1004 (http://www.cisco.com/c/dam/en/us/solutions/collateral/enterprise-networks/unified-access/cube-asr-release-10-0.pdf) but that's not within budget.  I've run it past a few people within Cisco and they have suggested it 'should' work as the features that are required (eg. SRTP, SIP TLS) are supported on CUBE but I haven't been able to find much evidence of someone actually having it working.

Curious to hear if anyone has (or hasn't) got it working.

Thanks,

Jonathan

[Jonathan Schulenberg]

Cisco tested with the ASR 1000 Series because that's what Microsoft lists as supported on their side. I agree with the others that it "should" work since none of the CUBE configuration in that document is unique to the ASR platform. Having said that, I also would never design/sell an ISR 2901 for CUBE since the CPU is so small. For example, a 2901 is only capable of one (1) call per second (see table 7 on page 15).

Also, I would put a firewall, use IOS ZBFW, or at least impose an ACL on the outside interface of CUBE to only allow SIP/RTP/SRTP traffic from the Exchange Online IP addresses. Remember that CUBE is implemented entirely in CPU, not an ASIC, and routers typically have a far weaker CPU than your modern x86 server.

Lastly, don't forget to monitor the utilization - both upstream and downstream - of the internet link that CUBE is going to transit out to Exchange Online. If you get reports of corrupted or distorted voicemails the most likely cause is the VoIP call being disrupted and the internet edge/uplink is the most likely place for this to occur.

[jonathan.dixon]

Thanks for the reply Jonathan, all great points.  I ended up testing with a C2901-VSEC-CUBE/K9 and it works but I'd agree the 2901 is limited in terms of performance.

The final config is quite similar to the Cisco guide for the ASR1004, you just need to add a secure transcoder to convert between RTP <-> SRTP.  Once that's in place and the certificates are set up correctly it worked fine, tested leaving and retrieving messages, MWI, play on phone, all working.

Regards,

Jonathan

[jsheehy13]

Jonathan - Were you able to get the auto attendant working to transfer calls or dial by name when searching.  I have this working as well using a Cisco 4331 ISR, but whenever SIP REFER is involved...the receiving phone starts ringing, but when you pick up the phone...it is fast busy.

[jonathan.dixon]

Hi jsheehy13 - Yes I just tested a basic dial-by-extension AA, it works fine, as you say SIP REFER is used so you need to have this enabled under 'voice service voip'.  My config originally had 'no supplementary-service sip refer' which caused no ringback but the call still worked.  Configuring the following fixed the no ringback issue:

!
voice service voip
 supplementary-service sip refer
!

That's the only difference between my 'voice service voip' config and what's in the Cisco guide (except the ip address trusted list where I entered all the address ranges from the Exchange Online IP addresses list linked above in Jonathan Schulenberg's post).  If that doesn't help have a look at codec negotiation and the output of 'debug ccsip messages' on the router.

Regards,

Jonathan