[Asterisk-Users] Sip behind the NAT

[chawki hammoud]

Hi list:
i have an asterisk box behind the NAT ,when i try to
send calls through Sip to the voip provider server the
call is answered but in a one way calling,I hear  the
voice of the other side just for 4 seconds and then
stop but the call do not hangup.

my sip.conf is:
[voip provider]
type=peer
host=213.112.50.8
username=XXXXXXX
secret=XXXXXX
fromuser=XXXXXXX
canreinvite=no
nat=yes
insercure=invite
disallow=all
allow=gsm
   


		
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