I cant make inbound calls in ucm6510 any help



i cant make inbound calls in ucm6510


This is not a very informative question. We have no idea of your telephone service type, the inbound rules, where you want the call to go, etc.

As a result, it is not possible to venture a guess as so little was made known.


Am using grandstream ucm6510 and the outbound calls are or,i can register with the sip provider but i cant call in,any help is helpful,i know i the topic might not be the best but u can recommend something


IT is likely related to the inbound rules and how formatted with regard to the DID seen by the PBX in the inbound INVITE.

YOu need to take a network capture and look at the URI or TO header for the DID. Then in the trunk set the DID mode to URI or TO, whichever you like, and then in the inbound rule, match the DID format exactly. So if the format is +988887765432 in the URI header, then you set the trunk to use URI and then copy +988887765432 into the DID spot for the inbound rule and set a destination.


above are my credentials
tel URI =Disabled
DID mode = To header
my pattern in inbound route is +256XXXXXXXXXX(couldnt type the number)
but nothing has worked and i cant see anything in CDR too,am looking forward to your feedback,thank you


Because the pbx registers and outbound calls work so apparently i thought i would just use the pattern as +256XXXXXXXXXX(the number given by the sip provider) and it would automatically have to work you can further comment on this please


Please capture and post


okay please!



My ucm6510 is,i filtered sip logs


tel%20uri ![did|471x34]





Waiting on your reply that is how i have configured,thanks


Your trace only shows communication between you Internal Ip phone and your pabx … He needs to see the Inbound call has it hits the pabx ( the pbx will receive the invite form the sip provider)



Even outbound test fine,i cant recieve anything on my pbx for inbound,you cauld advise on maybe setting i might be missing

Are my DID mode,TEL URI all okay? because that is the number the voip provider is giving me


Anything you can recommend?


The capture needs to be made when an incoming call is being made.


That is what i did but i dont capture anything on wireshark,i must be missing something that am not sure of,please check my screenshots i shared earlier and advise if i hv to make some changes in sip setting,trunk setting anything,thanks