asterisk users - May 2005 - Problems in new implemenation....

I have recently implemented a SIP VoIP implementation using Asterisk.  I can
go through and place a call to a particular number from the PSTN, the phone
rings, but I am not getting the ring response back to the calling party.  I
am not sure as to where this problem is coming from, but I know it stopped
working once I added the configurations....

dial-peer voice 82010151 pots
 incoming called-number 2010151
 direct-inward-dial
 forward-digits all
!
dial-peer voice 2010151 voip
 destination-pattern 2010151
 session protocol sipv2
 session target ipv4:XXX.XXX.XXX.XXX
 session transport udp
 incoming called-number 2010151
 dtmf-relay sip-notify rtp-nte
 codec g711ulaw
!
dial-peer voice 82010152 pots
 incoming called-number 2010152
 direct-inward-dial
 forward-digits all
!
dial-peer voice 2010152 voip
 destination-pattern 2010152
 session protocol sipv2
 session target ipv4: XXX.XXX.XXX.XXX
 session transport udp
 incoming called-number 2010152
 dtmf-relay sip-notify rtp-nte
 codec g711ulaw
!
!
sip-ua
 max-forwards 15
 retry invite 10
 timers trying 1000
 timers expires 300000
 sip-server ipv4: XXX.XXX.XXX.XXX
 no transport tcp
!

I also have a Cisco Call Manager Express sending and receiving calls to and
from this same equipment without the problem existing.  I am sure that this
problem is something with the way that I have the SIP commands configured on
this AS5400, but I just do not know enough to fix it.

Thanks for your thoughts.

Stephen

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