In/Out Calls - 1-2 Second Delay before caller can hear


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).

Any ideas?


What makes you think the problem doesn’t lie with your ITSP?

Note that canreinvite is deprecated. The correct name for this is directmedia.

Rather than guessing, you need to get a timestamped trace of the SIP and SDP. Asterisk can do this, but you must do it to logfile, not the console, for full timestamping, and you should enable millisecond resolution timestamps.