[Asterisk-Users] SIP / GrandStream Configuration

[WipeOut .]

Try adding nat=yes to your config..

Also if you want to make SIP to SIP extension calls and don't want to fight with the NAT set canreinvite=yes to canreinvite=no..

Finally set dtmfmode=info for the GS phones..

Later..

> 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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