I am trying to setup a Asterisk server using 2 network cards for the first time. To explain the setup:
used for all internet traffic and all sip devices will connect here
This is a private connection to a SIP provider and is used as the trunk.
So essentially when i make a call it is from a sip phone connected through eth0 and the call will go out through the trunk connected to eth1.
The problem i am having is that when i make a call it takes about 10 seconds before it even starts to call and when it does get connected i can only hear audio that goes from my SIP phone to PSTN. No audio is passed from pstn to my SIP phone.
I have tried set bindaddr = 0.0.0.0 in my sip.conf. As both networks are set up with public IPs i have nat=no . I am using asterisk 188.8.131.52. Would anyone know of what else it could be? Should asterisk be listening on both interfaces already or should i configure some IP rules to bridge the connections?
Thanks for any help.