I’m using AsteriskNOW version 11 and I had been fighting a strange problem that I finally managed to fix but I’m not sure why. I have two DIDs registered with Flowroute (SIP trunk) and both point to my Asterisk box and the Asterisk box turns around and forwards the call to two different external numbers over that same SIP trunk depending on which DID the user dials. For some reason on one DID when the call would come in and get forwarded to the external number the callers were able to hear each other perfectly. The other DID would forward the call but neither party had audio. After checking the configuration on Asterisk I found that the only difference was the inbound route for the DID that worked had a 1 second pause before answer and the other did not so I added the 1 second pause to the second DID and that solved the issue.
My question is why would the one second pause allow audio between the caller and called party?