Fwd: [Asterisk-Users] Nat Transversal

Zoa zoachien at securax.org
Mon Aug 1 10:54:59 MST 2005


Have a look at this tutorial about SIP and NAT problems, it might help
you...
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html


Zoa

David Romero wrote:

> now i forward ports 10000-20000 to my asterisk server but the problem
> is the same.
> i not understand wy not work
>
>
> ---------- Forwarded message ----------
> From: *Holden Hao* <holdenhao at gmail.com <mailto:holdenhao at gmail.com>>
> Date: Jul 28, 2005 9:09 PM
> Subject: Re: [Asterisk-Users] Nat Transversal
> To: David Romero <romdav at gmail.com <mailto:romdav at gmail.com>>
>
> On 7/29/05, David Romero <romdav at gmail.com <mailto:romdav at gmail.com>>
> wrote:
> > i try whit other codec but not work.
> >
> >  i try the phone on other site,
> >  and work nice just one time, i not change anyting and reboot the phone
> >  after reboot not work anymore,  if change the public ip address of my
> > router the phone work again just one time
> >  how i can fix it?.
>
> Your Asterisk has a private IP, right?  Check that you have properly
> forwarded all the ports  required.  Apart from 5060 for SIP your need
> to port forward the RTP ports 10000-20000 to your asterisk server.
>
>
> Holden
>
>
> --
> David Romero
> ##################################
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 254 bytes
Desc: OpenPGP digital signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/89a0c115/signature.pgp


More information about the asterisk-users mailing list