Yes, that is the solution. I have to set nat=yes in sip.conf. THX
-------- Original-Nachricht -------- Datum: Fri, 29 Dec 2006 15:10:47 +0800 Von: Dinesh Nair <[EMAIL PROTECTED]> An: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Betreff: Re: [asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1) > > > On 12/29/06 06:04 Hans-Jürgen Brand said the following: > > Found problem > > > > xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But > I don't know how to change this at xlite???? > > have you tried nat=yes in sip.conf for the peer ? > > -- > Regards, /\_/\ "All dogs go to heaven." > [EMAIL PROTECTED] (0 0) > http://www.openmalaysiablog.com/ > +==========================----oOO--(_)--OOo----==========================+ > | for a in past present future; do > | > | for b in clients employers associates relatives neighbours pets; do > | > | echo "The opinions here in no way reflect the opinions of my $a $b." > | > | done; done > | > +=========================================================================+ > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
