I have two machines on the internet. Box A and Box B.

Box A has a SIP trunk to the world, Making calls box A works fine I have audio to my cell and all works.

I defined a SIP trunk between box B and A. tried to make a call originating from box B – to box A and then over the SIP trunk to my cell.

My cell rings but then no audio.

I have defined SIP trunks before between boxes pretty straight forward. I have checked and my firewalls are open for SIP/RTP
-A INPUT -m state –state NEW -m udp -p udp –dport 5060 -j ACCEPT
-A INPUT -m state –state NEW -m tcp -p tcp –dport 8000:60000 -j ACCEPT
-A INPUT -m state –state NEW -m udp -p udp –dport 8000:60000 -j ACCEPT

I am using asterisk 11.16

box A is
[boxab_sip]
type=friend username=boxa_sip secret=***
disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833
host=DNS Name here context=sip_trunk insecure=port,invite

box B is
[boxab_sip]
type=friend username=boxab_sip secret=***
disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833
host=DNS Name here context=sip_turnk insecure=port,invite

Is there something I am missing?
The one piece I have not done before is SIP trunk – to – SIP trunk. But the phone rings – so its routed – just no audio.

Thoughts?

Thanks,

Jerry