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Trouble with inbound calls in CUCM 6.x

fieryhail
Level 1
Level 1

Hello all,

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.

2 Accepted Solutions

Accepted Solutions

asandborgh
Level 4
Level 4

Hi there,

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

direct-inward-dial

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.

HTH,

Art

View solution in original post

Hi,

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

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b03htpil.html

Paul

View solution in original post

9 Replies 9

Paul Reck
Level 1
Level 1

Hi there,

if you've got an MGCP gateway and can dial out, check the CUCM gateway FXO configuration and ensure that it has a Calling Search Space that contains the partition that the DN you're trying to reach is in.  If not then the gateway won't have the rights to reach the phone.

The route patterns/groups/lists tend to be used mainly for outbound calls and so aren't going to be affecting this inbound issue, but a great place to do some reading on how to best set them up is in the SRND

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/dialplan.html

Paul

asandborgh
Level 4
Level 4

Hi there,

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

direct-inward-dial

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.

HTH,

Art

Just re-read your message - never mind about the DID (I blew by the FXO).  Don't know why you would be getting second dial tone since an FXO port will generally not provide it.  Do check the Attendant DN and move your DN opn the phone to the none partition.

Thank you all for the quick responses.  I put the DN (1001) in the partition.  I also put my extension (1001) in the Attendant DN on 4/0/0 (one of the FXO ports).  I am not sure what is going on, I dial in and get the message about "The call can not be completed as dialed, please consult your directory."  That message proves that UCUM IS handling the call because I know it is NOT a message served up by my PSTN provider.  I'm honestly not sure where the problem is.  I do not have any scurity measures included currently, just trying to establish basic call connectivity, then add voicemail services.  Any ideas?  Thanks again.

Thank you for the help.  After resetting the gateway, now it does work.


Not sure if I should open a new thread, but does anyone have knowledge on how to make multiple extensions ring simultaneously when a call comes inbound?  As opposed to having the DN of a single IP phone, maybe there is a way to create a DN that will ring multiple other DNs at the same time?  Thanks again.

Hi again,

did you check on Art's good suggestion about setting the number of expected digits on the gateway to match the number of digits you're expecting? (4 in the case of 1001)

Paul

Edit- ahh, I misread initally but I see you've got it working now, nicely done

Hi,

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

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b03htpil.html

Paul

Thank you very much for the response Paul.  I think the Shared Line is what is most appropriate for this scenario.  I will be reading through the material in this link also.  Very good information, thank you.  I believe I will try setting up a broadcast hunt group.  I'll write back to let you know how it goes.  If you have time, do you know where I might find information on creating a shared line?  Sorry for the naive questions, but I'm still very new to CUCM.  Thanks again in advance.

I have read through that information, I believe I would want to setup a hunt pilot such as 100X to ring DNs 1000-1009 at once.  Now what I can't seem to figure out, is how to apply this hunt pilot so that it "answers" when a call comes in the gateway.  I've configured the Hunt Pilot, List, and Group.  Now just can't figure out how to have that hunt group answer when a call comes in on either FXO port.

Just wanted to say Thank You sincerely to those of you who helped me out.  I figured out the broadcast hunt group settings.  I know I have a lot more work ahead of me.  The next thing will be to setup the connection for voicemail with Unity Connections, as well as IVR functionality.  Once again, thanks!

No problem at all,

Good luck with the Unity Connection setup, there's plenty of information out there in the Cisco design and administration guides to get you started but if you run into difficulty Netpro is a great resource.  Have a search around as there are rarely problems you'll come across that someone hasn't come across before and documented here.  If not there'll be someone willing to help you out.

Paul 

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