[Asterisk-Users] SIP / GrandStream Configuration

[Uriel Carrasquilla]

Hi there!
I installed the BudgetTone (GrandStream) on my LAN without any problems.
Then, I moved it to another location using a D-Link NAT.
I opened 5060 (SIP) and 5000 to 5008 for RTP.  I also fixed the IP address
of the BudgetTone.
When I receive a call on my Asterisk, it would ring my FXS as before.
However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in
the log).
The configuration I  have in * is the following:
sip.conf
-----------
[general]
port=5060
context=sip
maxexpirey=3600
defaultexpirey=60
disallow=all
allow=ulaw
allow=gsm
[1000]
contet=sip
type=friend
username=1000
secret=?????  (not the real one)
host=dynamic
mailbox=1000
canreinvite=yes
dtmfmode=rfc2833

I did not change the above configuration when I moved the budgetTone from
the LAN to the Internet (Wan).
I am not using a "register" statement in the sip.conf and I am wondering if
I need to.
I did change the sip server IP address in the Grandstream configuration.

I suspect my problem is with the router (NAT).  I don't quite understand the
symetric discussions but I downloaded a paper to learn more.  Right now, all
my public and private ports are the same.

Regards,
Uriel

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