I am having the following problem. I have 2 real ip addresses and a address with heartbeat.
I have asterisk 1.2 with canreinvite=no in sip.conf and it worked fine. Then I upgrade to 1.4 and can’t hear when I call but the called party can hear. The weird thing is that rtp debug shows no packet arriving but tcpdump shows packets arriving and leaving. Bind address is heartbeat address.
any help would be appreciated.