I am trying to get a CUCM Business system up and running. This is currently a test environment. Am using a 3745 with NM-2V and a VIC-2FXO as my voice hardware. CUCM 6.01 Business edition on ESX server. I know it is an unsupported config, this is still test phase before laying out for an MCS or compatible server. I have worked more with the Asterisk/FreePBX systems so CallManager is new to me. I am unsure of how to get calls initiated from the PSTN to route to Cisco IP phones inside. I can dial out to the PSTN from an IP phone, but when I try to dial in I either get a second dial-tone or the error message, "Your call can not be completed as dialed, please check the number or consult your directory." I think it has something to do with partitions and Calling Search Spaces along with the associated route patterns. Can anyone give me direction as far as this goes? Perhaps some examples on a basic setup of partitions/CSS as well as a route pattern/list that would allow inbound and outbound calling? I didn't mention it earlier, but I am using MGCP as the gateway protocol. Trying to anyway. I'm open to using h.323 if that is thought to be the better option. Thanks again for your assistance.
There are a couple of ways of multiple extensions ring simultaneously. You can configure a shared line, so multiple phones have the same DN associated perhaps on a second line of the phone or a use a broadcast hunt group.
For a hunt group you configure a hunt pilot, a hunt list and a line group containing the DNs that you want to receive the broadcast. The hunt pilot reference the hunt list, the hunt list contains line groups and the line groups contain the DNs. There are quite a few options available and it worth have a read through the following guide
This really shouldn't apply, since you are running MGCP, but try it anyway. Second dial tone is usually caused by no DID dial-peer on a digital trunk. Input the following:
dial-peer voice 1111 pots
incoming called . (note the period)
Also is your inbound trunk an FXO, a PRI/BRI, CAS?
If it is an FXO make sure you have the number of your IP phone in the "Attendant DN" field so that the incoming call is routed there. If t is a PRI/BRI you need to set your "expected digits" field on the gateway to trim the number down to your extension length.
Additionally, for the purpose of testing and getting the call in, DON'T put the DN you are using on the IP phone in ANY partition/CSS. If it is left in the "none" partition everything can see and access it, so you have rempoved that variable.