asterisk users - Feb 2015 - SIP trunk no audio

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150218/36449dd0/attachment.html>

> But the phone rings - so its routed - just no audio.
The ringing is SIP signaling.  The audio is RTP data.  See if the audio 
is getting routed with a sniffer.  Maybe use one codec that both clients 
support.

Adrian Serafini