Adam: I believe you. I assume that the RTP is creating a symetric configuration between * and the SIP phone. The situation we are left to live with is that * (won't be the Sip phone) can only live in the Internet brave world (and not behind a firewall). is this acceptable? Uriel
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Varga Sent: Wednesday, September 24, 2003 11:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
