[Asterisk-Users] SIP / GrandStream Configuration

[Brancaleoni Matteo]

have you tried to put nat=yes in the user definition in sip.conf ?

Also, the * server is on a public IP?

Matteo

Il mer, 2003-09-24 alle 15:35, Uriel Carrasquilla ha scritto:
> 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
>  
-- 
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl