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Hi, 

 

We are moving from a Cisco TMS/VCS/Codian environment to a Polycom DMA/RM environement and one area that is causing me grief is how to handle my SIP voice calls to our DMA managed virtual meeting rooms.

 

Internal calls are not a problem, because we can handing the routing on call manager and allow people to just dial the VMR on their phone. The problem is calls coming in externally.

 

Prviously we had all incoming external audio calls for VC's land on a Codian ISDN gateway, where the auto attendant would handle the call routing to the correct VMR on the MCU. We moved away from thsi about 6 months ago as we intended to retire the gateway and the ISDN services. In it's place we had a directory number set up on Call Manager, that would route to the Cisco VCS, the VCS would perform a translation that would then match a rule to send the call to the auto attendant on the Codian MCU.

 

Now, if we were just looking to continue using the one MCU we could do the exact same thing with the DMA, but one of the main drivers for going with the Polycom solution was the MCU pooling, as we can no longer route the calls to the MCU because the MCU isn't doing the conference set-up, and with the MCU pooling it may not even be the correct MCU.

 

Now I know that the DMA features shared number dialling and virtual entry queues but these only work with RMX's. Yes, we are deploying a RMX as well, but our other two MCU's will be Codians.

 

We can go back to using the ISDN gateway, but we want to get away from that if possible. 

 

So what are our options? Has anyone else faced a similar challenge? Any suggestions?

 

I'm guessing we'll need a SIP voice gateway with an auto attendant function? What should I be looking for?

 

Any comments, suggestions or recommendations are welcome!

 

regards

 

Shane

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