no audio on end-point when call is connected/bridged via PBX

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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@carrier.callwithus.com
register => xxxx:31@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()