Try setting canreinvite and nat to no for those extensions. Saludos, Juan E. Rodríguez
-----Original Message----- From: Alyed <[email protected]> Date: Fri, 26 Mar 2010 10:56:50 To: Asterisk Users Mailing List - Non-Commercial Discussion<[email protected]> Subject: Re: [asterisk-users] need help on setup rtp directly between 2 sip clients -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
