Hello,
Established a SIP trunk from Asterisk (IP = x.x.x.x )with an operator MSC (IP = z.z.z.z) and routed a number from operator side to our Asterisk platform for test purposes.
When we are making a test call, in dump file can see only INVITE, BYE, ACK from z.z.z.z to x.x.x.x but cannot see any answers back from our platform x.x.x.x IP, however the calling party was able to hear the test prompt so calls are successfully
Maybe there are some configs on asterisk that can filter the dump or other reasons, did anyone face some similar issue?