[Asterisk-Users] SIP / GrandStream Configuration

[Uriel Carrasquilla]

Michael:
could you share how you configured your GrandStream?  for example, did you
say "yes" to NAT (without a STUN)?
how about in SIP.CONF, how did you configure the remote GrandStream?
Regards,
Uriel
  -----Original Message-----
  From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Michael Koehler
  Sent: Thursday, September 25, 2003 10:42 AM
  To: asterisk-users at lists.digium.com
  Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration


  Sorry, but my * is behind NAT and i have no problems with SIP, and it even
works with NAT to NAT and without forwarding ports or similar effords.


  Michael


  Stephen Varga wrote:

On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote:

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


Nope, it is not currently possible. * behind a NAT for SIP does not work
because the * real IP address is placed in the SDP information,
therefore the 'outside' phone can not send the media stream to *. See my
answers over the last week for the more details and possible work
arounds.

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users




-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/0efb09d3/attachment.htm