Re: [Asterisk-Users] SIP / GrandStream Configuration

[Michael Koehler]

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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users