have you been using asterisk and maybe is this your first sipXecs system?

On Fri, Jul 8, 2011 at 7:07 AM, <[email protected]> wrote:

>  I don't use domain names in my VoIP system. Only IP's..
>  That's why I don't think that the reason of the problem is DNS, maybe
>  I'm wrong?
>  Regards, kga.
>
>  On Fri, 8 Jul 2011 07:02:03 -0400, Michael Picher <[email protected]>
>  wrote:
> > Have you done your DNS tests remotely (in the same manner a phone
> > might)?
> >
> > On Fri, Jul 8, 2011 at 4:48 AM,  wrote:
> >   Hey there!
> >   This is my first post in sipx-users mailing list, so I'm sorry if
> > I do
> >   something wrong.
> >   Well.. the problem is strange enough, and I can't find out how to
> > fix
> >   it during 2 days, hm..
> >   So, i've got a sipX station, and it has public IP. So "Server
> > behind
> >   NAT" is unchecked.
> >   There are about 5 users, whos phones use NAT. Everything is ok
> > with
> >   softphones and SIP mobile phones. The only settings i fix in the
> >   softphones and SIP mobiles phones are REG server and
> > Domain/Outbound
> >   Proxy server. Everything works fine! I use this pretty good guide
> >
> >
> >  _http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal
> > [2]
> >   But! The only phone (Siemens Gigaset C470 IP) doesn't work as good
> > as i
> >   need. The scheme of my VoIP network looks like this:
> >   SIPX Server - > IP-network -> Remote NAT + Siemens Gigaset C470 IP
> >   So the sipX station does'n use any nat, but the phone does. SIP is
> > ok,
> >   so I can call this user and he can call me. When the call goes
> > from
> >   Siemens Gigaset C470 IP we can speak and everything is ok. But if
> > I call
> >   him, there is no media between our phones. So the remote user can
> > hear
> >   the ring, pick up the phone, but we can't hear each other..
> >   I've already tried to fix everything I could. There are those
> > types of
> >   options in the Siemens Gigaset C470 IP web GUI:
> >
> >   Domain
> >   Proxy server address and port
> >   REG server addres and port
> >   STUN yes/no
> >   STUN server
> >   Outbound proxy server and port
> >
> >   I've alredy tried different combinations of those settings (even
> >   checking "use STUN"), but there is no result. No media when anyone
> > call
> >   him.
> >   I will be happy if someone could help me to find out where the
> > problem
> >   is.
> >   Regards, kga.
> >
> >  _______________________________________________
> >  sipx-users mailing list
> >  [email protected] [3]
> >  List Archive: http://list.sipfoundry.org/archive/sipx-users/ [4]
> >
> > --
> > Michael Picher
> > eZuce
> > Director of Technical Services
> > O.978-296-1005 X2015
> > M.207-956-0262
> > @mpicher
> >  www.ezuce.com [6]
> >
> >
> >
> > Links:
> > ------
> > [1] mailto:[email protected]
> > [2]
> > http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal
> > [3] mailto:[email protected]
> > [4] http://list.sipfoundry.org/archive/sipx-users/
> > [5] http://twitter.com/mpicher
> > [6] http://www.ezuce.com
>
> _______________________________________________
> sipx-users mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



-- 
Michael Picher
eZuce
Director of Technical Services
O.978-296-1005 X2015
M.207-956-0262
@mpicher <http://twitter.com/mpicher>
www.ezuce.com
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/

Reply via email to