05-12-2016 05:37 AM - edited 03-17-2019 06:53 AM
I have Unity system where I am trying to setup a call handler. As part of the call handler I want the outside parties to be able to dial a person's extension if they know it. If the person is on Unity and they dial the person's extension it takes them straight to their voicemail stating the person is unavailable. If they dial someone that is not on Unity then you get an error stating the call cannot be completed.
I checked the system restrictions table and made sure the last item is not blocked. I also made sure the Allow Transfers to Numbers Not Associated with Users or Call Handlers is checked marked.
I think the underlying issue is that my Unity system is not able to dial out to call manager. I have integrated my CUCM and CUC with a SIP trunk.
If you can provide some assistance on how to best troubleshoot this that would be great.
Thank you in advance.
Solved! Go to Solution.
05-12-2016 09:39 AM
Alongwith valueble suggestions given below, i would also make sure the inbound calls CSS on the unity SIP trunk contains the partitions for all users..
Cheers!
Deepak Mehta
05-12-2016 07:10 AM
Hi,
I will start with looking at SIP traces in CUCM and check if call is hitting CUCM or not.
- Vivek
05-12-2016 09:51 AM
I will check that now.
05-12-2016 12:46 PM
Yes, I think you are correct, how would I go about correcting that issue?
05-12-2016 01:15 PM
Do you need help in looking through the SIP trace .
if yes then go to RTMT-Real time session trace set the filter for 30 mins and then check which calll was the failed call or you will have to collect the CM traces.
Also check the CSS for inbound calls at below in SIP trunk config.
05-12-2016 09:17 AM
05-12-2016 09:50 AM
I checked that and the rerouting CSS was set to None, I changed it to the correct CSS and still no go. As an FYI, I am not trying to get out to the PSTN, I am trying to get back to extensions on my CUCM.
05-12-2016 09:39 AM
Alongwith valueble suggestions given below, i would also make sure the inbound calls CSS on the unity SIP trunk contains the partitions for all users..
Cheers!
Deepak Mehta
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If you have checked the Restriction Table and that looks fine then I would like you to check the below things:
Check if the SIP trunk's Rerouting CSS had the partition of the Route Pattern to the PSTN number. If the call is transferred via a CTI RP/Translation pattern - The voicemail port/SIP trunk must have access to it and the CSS of CTI RP/Translation pattern must have the partition of the RP to the PSTN number.Regards
Deepak