Hi Community, hoping to get some help on my issue.
We have a call flow that routes calls from CUCM to Microsoft Teams Voice Direct Routing (SBC) then out to PSTN. The main number is configured in CUCM as a translation pattern to translate into a new number.
(Translation pattern in CUCM routes to a number configured in Teams Admin Center which uses a VRP to route to a SBC which sends it out the the PSTN)
The call is successfully transferred to the SBC where it rings the end device. However the call is prematurely canceled before the device answers.
In the SBC logs (ribbon LX) I see a call clearing code in the SIP messages (Reason: Q.850; cause=47). I looked up the code and it refers to “codec mismatch”.
How would I resolve?
Thank you in advance.
Using: CUCM 10.5.2 | SBC Ribbon | using E.164 formats
Please explain where FreePBX (or even Asterisk) fits in here.
However, I would note that CUCM often uses late offer SDP, which means the codec compatibility issues will not be discovered until after the called party has answered, so it is quite possible for there to be ringing before CUCM discovers the call isn’t viable. (You haven’t said which party initiated the clear, and whether it was BYE or CANCEL.)
(It has something to do with the use, or non-use of MTPs, but I forget the exact reasoning. If you don’t have MTPs (and I think if there are no free MTPs) both parties must share a codec, as CUCM uses MTPs to transcode.)
Long time ago had similar issue (not was an asterisk appliances) and I remember CUCM administrator set a separated Media resource group As Media Resource provides, Conferences Bridge, Transcoding, Media termination point and music on hold.