Hello, I have 2 sip trunks from 2 different providers, both are directly connected to my asterisk PBX(through a p2p connection with the telcom) which acts as a gateway and routes calls accordingly.
On the FreePBX side, I set up a conference bridge and an inbound route which comes from asterisk whose destination is the conference room.
from one of the sip lines, everything works well, but the other connects, but no audio can be heard, not even the greeting which asks for the pin, but normal incoming and outgoing calls work well, it’s only when I trying to access the room that no audio is heard and the call does not drop. I have tried allowing anonymouse and guest calls, but still cant get the audio to work