asterisk users - Dec 2010 - no audio on end-point when call is connected/bridged via PBX

I am trying to dial through my asterisk machine from phone A to phone B.
My DID is registered properly with the SIP provider.  When I dial from
A to B it looks fine so far.
A rings B and B can pick up and the call is bridged.  However, I don't
hear any audio so therefor it is not working.
I am running Asterisk 1.8 on a cloud server.  I have had the same
configuration running on a physical machine with a similar
configuration.
Thoughts?  I know I posted this yesterday but was hoping for some more
creative comments!

Zip*CLI> sip show registry
Host                                    dnsmgr Username       Refresh
State                                                    Reg.Time
sip.callwithus.com:5060                 N      xxxx          105
Registered                                               Tue, 07 Dec
2010 02:36:43
1 SIP registrations.

my sip.conf
[general]
context=default
allowoverlap=no
;bindport=5060
port=5060
bindaddr=0.0.0.0
canreinvite=no ;if your asterisk box is behind a NAT ro

;register => xxxx:31 at carrier.callwithus.com
register => xxxx:31 at sip.callwithus.com

[callwithus]
type=friend
host=sip.callwithus.com
username=xxxx
secret=31
qualify=no
insecure=invite


my extensions.conf
[general]

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus

[default]
exten => s,1,Answer()
exten => s,n,Dial(SIP/callwithus/12222222222)
exten => s,n,Wait(2)
exten => s,n,Hangup()

> I am running Asterisk 1.8 on a cloud server.  I have had the
> same configuration running on a physical machine with a
> similar configuration.
> Thoughts?  I know I posted this yesterday but was hoping for
> some more creative comments!
If signalling works and audio don't, it probably has to do with phones
behind NAT. It seems necessary to review the configuration of local routers.

Regards,
Jos? Miguel

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