[Asterisk-Users] SIP / GrandStream Configuration

[Uriel Carrasquilla]

Adam:
in reference to my first message, the NAT on the SIP/GS (a D-Link router)
has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being
forwarded to the Sip/GS.
The Asterisk server, also behind another NAT (Linksys), has the same ports
opened and forwarded.
is it still impossible?
URiel

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Adam Hart
Sent: Wednesday, September 24, 2003 7:27 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration


How will the packets get to the asterisk server? You'd need to forward ports
on the NAT device, otherwise it's impossible

----- Original Message -----
From: "Uriel Carrasquilla" <uriel at adelphia.net>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, September 25, 2003 9:48 AM
Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration


> Very valuable help.  It is now working like a champ.
>
> This is a solution with SIP--NAT---Internet---Asterisk.  No problems here.
>
> What I would like to do next is to move Asterisk behind a NAT as follows
> SIP---NAT---Internet---NAT---Asterisk
> do I need a STUN server? is there a chance this could work?
> The Google results seems to indicate that I will get an ulcer attempting
> this step.  is that true?
>
> Regards,
> Uriel
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of WipeOut .
> Sent: Wednesday, September 24, 2003 9:05 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration
>
>
> 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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