I have an issue on Asterisk 1.8 Cert with 1-2 seconds delay before I can hear the destination.
So, usually I cannot hear “Hello” or “Welcome” (“Welcome to” if it is Automated Attendant) from the destination when I call to outside. When I pickup the phone to answer a call (incoming call), I have to wait 1-2 seconds before the caller can hear me, but at the same time (1-2 sec) I can hear initiator.
So, I use SIP trunk adn Asterisk server has public IP address.
sip.conf contains setting canreinvite=no (RTP passes via Asterisk server as well).