Hi, I’m a new asterisk user and I need some help with my setup and i hope someone can point me in the right direction.
I’ve just installed asterisknow and i’ve two users (ext 7000 & 7001) configured. I’m able to connect to asterisknow using SIP softphones and can call between the two extension.
Next i tried to connect to a Cisco 2811 router with an E1 to the PSTN using H323. I configured a dial plan on asterisk to forward all PSTN calls (_9X.) to the H323 trunk and configured the necessary dial-peers on the router. The problem I have is when i tried to place a call from the asterisk phones to the PSTN or from the PSTN to the asterisk phones, the phones rings but when the call is picked up, there is no audio??
Am I missing anything here?? Is there anything i need to do on *now to permit SIP<>H323 calls? My H323.conf is as follows,
port=1720
bindaddr=0.0.0.0
disallow=all
allow=ulaw
dtmfmode=rfc2833
gatekeeper=DISABLE